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| 00:35.23 | Cypris | i have a Panasonic KX-TGP-550 with a kx-tpa50 handset, im trying to setup the handset with its own extension but asterisk is not allowing me to create an extension using a mac address thats already in use, anyone have advise on this? |
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| 00:39.48 | WIMPy | That question doesn't make much sense. Are you using some GUI that makes that kind of relation? |
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| 00:47.54 | [TK]D-Fender | Asterisk does not care about MAC. |
| 00:49.14 | Cypris | i suppose i am using trixbox but still. this is making my head hurt |
| 00:49.59 | [TK]D-Fender | Trixbox is massively dead |
| 00:50.03 | [TK]D-Fender | like 4 years dead |
| 00:50.52 | [TK]D-Fender | Anyway... shouldn't stop you from using more appropriate terms and a more complete description of what you're actually doing to try and get it running |
| 00:51.14 | [TK]D-Fender | But as a note that server is a massive security liability |
| 00:51.58 | [TK]D-Fender | So * does not ack about MAC anywhere. |
| 00:52.13 | [TK]D-Fender | You should probably tell us where you decided to plug that into |
| 00:52.21 | [TK]D-Fender | and .... he's off |
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| 02:06.22 | DruidZ | just did an emergency switch from FS to Asterisk. |
| 02:07.41 | WIMPy | What kind of emergency? |
| 02:07.43 | DruidZ | just did an emergency switch from FS to Asterisk. |
| 02:07.57 | DruidZ | FS fell down and wouldn't get up again. |
| 02:08.27 | DruidZ | Can someone tell me how to get a list of users who have registered? |
| 02:08.49 | DruidZ | I can see the list of users but I know that they aren't all registered. |
| 02:09.08 | DruidZ | There doesn't seem to be a CLI command for that. |
| 02:09.10 | WIMPy | sip show peers |
| 02:10.30 | DruidZ | So if the host is "Unspecified" that means unregistered? |
| 02:10.43 | WIMPy | yes |
| 02:11.11 | DruidZ | Ah. Thanks. |
| 02:13.49 | DruidZ | How do I get it to display "NAME/NUMBER" instead of "NUMBER/NUMBER"? |
| 02:14.07 | WIMPy | Where? |
| 02:14.10 | DruidZ | same => n,Set(CALLERID(num)=Somebody <== doesn't seem to do it. |
| 02:14.20 | DruidZ | in sip show peers |
| 02:14.22 | WIMPy | (name) |
| 02:14.50 | DruidZ | same => n,Set(CALLERID(name)=Somebody) |
| 02:15.01 | DruidZ | Sorry, C&P the wrong line. |
| 02:15.19 | WIMPy | that's neither name nor number. It's the username and the callbackextension. |
| 02:15.41 | DruidZ | Ah. Got it. |
| 02:16.22 | WIMPy | That's why the 2nd part is missing for unregistered peers. |
| 02:16.29 | DruidZ | Sorry if I seem needy. I spent the day learning Asterisk and setting up a production system. There are still a few holes. |
| 02:17.03 | [TK]D-Fender | What went so wrong with your FS you decided to change platforms entirely to something you don't know for production? |
| 02:17.04 | DruidZ | The biggest is voice mail. Need to research that tomorrow. |
| 02:17.26 | DruidZ | It just would not run any more. No idea why. |
| 02:17.40 | [TK]D-Fender | And attempts to fix it with the community? |
| 02:18.03 | DruidZ | It was always a little fragile. Asterisk already feels more solid. |
| 02:18.40 | coppice | Asterisk feels solid to someone who doesn't know how to use it? :-\ |
| 02:18.41 | WIMPy | Don't worry. That feeling will go away. |
| 02:18.47 | WIMPy | At least fo some time. |
| 02:19.07 | WIMPy | *for |
| 02:19.13 | DruidZ | Well, I can restart it without rebooting. That's a plus. |
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| 02:21.02 | DruidZ | I should have gone with Asterisk in the first place. |
| 02:21.38 | WIMPy | The grass is always greener on the other side. |
| 02:21.51 | WIMPy | I wish I hadn't started with Asterisk. |
| 02:22.54 | DruidZ | The only thing with FS was that voice mail worked out of the box. |
| 02:23.38 | [TK]D-Fender | If comes in a box? You know what else comes in a box? CORPSES |
| 02:23.58 | [TK]D-Fender | </macabreallegory> |
| 02:24.01 | WIMPy | LOL |
| 02:24.03 | jmmills | I don't remember, is there a way to execute a function on a channel from the console |
| 02:24.15 | jmmills | an active channel |
| 02:24.17 | WIMPy | no |
| 02:24.43 | WIMPy | YOu can only redirect it to some dialplan that does it. |
| 02:24.53 | [TK]D-Fender | No, one other way |
| 02:24.55 | [TK]D-Fender | AMI |
| 02:25.27 | WIMPy | Still the same. |
| 02:26.22 | [TK]D-Fender | No need to redirect |
| 02:27.15 | WIMPy | You're right. I was thinking about functions that might do something to the call, but I guess that's something that just doesn't exist. |
| 02:31.05 | [TK]D-Fender | Depends what you want to do as to what ideas are viable. |
| 02:31.24 | [TK]D-Fender | jmmills: So waht are you specifically looking to do? |
| 02:31.38 | jmmills | Just silly things like force a playback on a channel |
| 02:31.42 | jmmills | experimentation |
| 02:33.10 | WIMPy | If you wanted to ask abotu Applications instead of functions, then redirecting IS the only way. |
| 02:35.16 | [TK]D-Fender | ... sorta :) |
| 02:35.37 | WIMPy | Or use AGI from the start. |
| 02:35.41 | WIMPy | Anythign else? |
| 02:35.41 | [TK]D-Fender | [22:31]jmmillsJust silly things like force a playback on a channel <- AMI Originate, Chanspy (w), Playback |
| 02:35.49 | [TK]D-Fender | ^ |
| 02:35.58 | jmmills | yeah, but I was thinking from an active channel via cli |
| 02:36.02 | WIMPy | In that special case, yes. |
| 02:36.24 | [TK]D-Fender | Every case is special, but net every one gets its own little bus ;) |
| 02:36.29 | jmmills | You can set a variable on a channel while it's connected |
| 02:36.29 | [TK]D-Fender | not* |
| 02:36.44 | [TK]D-Fender | jmmills: You can't set a variable on a call that doesn't exist... |
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| 02:36.56 | [TK]D-Fender | Egg eats chicken, news at 11.... |
| 02:37.10 | jmmills | well, I just realized that setting DYNAMIC_FEATURE only works before the channel is connected |
| 02:37.31 | WIMPy | Makes sense. |
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| 02:56.58 | DruidZ | Are there issues with 3CX and Asterisk? |
| 02:57.30 | DruidZ | I can register my Linksys ATA but the two Androids with 3CX won't register. |
| 02:59.02 | [TK]D-Fender | No. |
| 02:59.08 | [TK]D-Fender | BRB |
| 02:59.19 | DruidZ | Oh, I set he server as the proxy server and it connected. |
| 02:59.32 | DruidZ | I wonder why I didn't need to do that with FS. |
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| 08:35.08 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.8.1 (2014/03/10), 1.8.26.1 (2014/03/10); Standard: Asterisk 12.1.1 (2014/03/10); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 11:07.05 | bitwize | Hi guys, I need some help with my AMI configuration. When I receive an incoming call I create a channel variable and populate it with a customer related identifier. I want this identifier to be passed through my AMI manager in every event related to the channel. I have configured the "channelvars" option in manager.conf to include the variable but it seems to be ignored. Only "standard" variables are being passed. I'm working on version |
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| 11:11.24 | vassilux | hi all, I use wheezy with rseries R850, after remove brltty package I can access to /dev/ttyUSB0 but the device /dev/rseries0 is not created Any idea ? |
| 11:12.30 | vassilux | It works well with squeeze , I have some clusters on squeeze boxes and work very well for a long time |
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| 11:49.45 | WIMPy | bitwize: Use the channel name to identify your channels. |
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| 12:47.07 | amessina | I'm running into a problem (also reported by another user here: http://forums.digium.com/viewtopic.php?p=196548) with Asterisk 12 and PJSIP TLS ciphers. No matter what I put for "cipher=", I get "Ciper is unsupported". I have built PJSIP and show that it properly finds OpenSSL libs during the configure. |
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| 13:06.58 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:07.12 | Ice_Strike | Does anyone know if Asterisk perform better on ESXi or Hyper-V? |
| 13:07.42 | Greenlight | I'd stick to ESXi all day long |
| 13:08.33 | Ice_Strike | Cool |
| 13:08.39 | Ice_Strike | We use ESXi at work ;) |
| 13:08.53 | Ice_Strike | Mainly two DC's and a internal web server |
| 13:09.01 | Greenlight | WE've a load of virtual Asterisk boxes running on ESXi without issues. |
| 13:09.01 | *** join/#asterisk DruidZ (~darcy@dilbert.druid.net) |
| 13:09.13 | Greenlight | You run two DC's on the same ESXi box ? |
| 13:09.17 | file | I develop under Hyper-V but don't use it in production. |
| 13:09.33 | Ice_Strike | Greenlight Sadly yes. |
| 13:09.53 | Ice_Strike | I like to get 1 more ESXi box and run as cluster or somehow |
| 13:09.55 | Greenlight | Doesn't that defeat the purpose of a second DC (eg redundancy) |
| 13:10.21 | Ice_Strike | Greenlight It does but if Windows Server DC1 messed up, it will use DC2 |
| 13:10.30 | Ice_Strike | Yea hardware is other story. |
| 13:11.20 | Ice_Strike | Greenlight Ever used vCenter Server? |
| 13:12.01 | Greenlight | Naa, we use vSphere for management |
| 13:12.20 | file | Greenlight, how's your install running these days? |
| 13:13.04 | Greenlight | Pretty perfectly atm - touch wood! |
| 13:13.11 | file | awesome |
| 13:13.14 | Greenlight | mhm |
| 13:13.46 | Ice_Strike | yea I use vSphere for management |
| 13:13.53 | Ice_Strike | using paid version or free? |
| 13:14.07 | Greenlight | Paid |
| 13:14.16 | Greenlight | I though you had to after a certain level |
| 13:14.23 | Greenlight | It's not much though |
| 13:14.43 | Ice_Strike | You can't backup VM's using free version |
| 13:14.55 | Ice_Strike | API is locked in free version |
| 13:15.25 | Greenlight | I seem to recall some other limitations, but can't remember specifically. I figured a few hundred quid a year was fine |
| 13:15.49 | Greenlight | Going to be doing some work with the API soon actually - want to auto provision virtual asterisk boxes for customers, and fully set them up |
| 13:16.08 | Ice_Strike | That would be fun |
| 13:16.16 | Greenlight | Like clone a VM from a template, and work from there |
| 13:16.21 | Ice_Strike | Yep |
| 13:16.32 | Ice_Strike | I think XEN is easier to work with. |
| 13:16.50 | Greenlight | Only really played around seriosuly with ESXi |
| 13:17.04 | Ice_Strike | How many asteriskVM's you have in a box? |
| 13:17.58 | Ice_Strike | Not sure if Mware VCenter Server woudld help... if asterisk VM goes down and then continue running from other esxi box |
| 13:18.05 | Ice_Strike | like failover |
| 13:18.07 | Greenlight | hmm maybe 20 or so |
| 13:18.14 | Ice_Strike | Nice |
| 13:18.17 | Greenlight | Plus a few windows boxes on there |
| 13:18.28 | Ice_Strike | Approx concurrents channels? |
| 13:18.36 | Ice_Strike | That must be a powerful box |
| 13:18.43 | Greenlight | More than 1000 |
| 13:18.49 | Ice_Strike | lol nice! |
| 13:18.50 | Greenlight | Perhaps pushing 2000 |
| 13:19.04 | Ice_Strike | Via a single network card? |
| 13:19.19 | Greenlight | Yea, it will be |
| 13:19.45 | Ice_Strike | No Lag issue? |
| 13:19.46 | Ice_Strike | Nice |
| 13:19.52 | Greenlight | No, all runs sweet |
| 13:20.02 | Ice_Strike | Lovely. |
| 13:20.45 | Greenlight | Most are customers boxes, but we've also a "billing gateway" which is a streamlined asterisk box that all call traffic for all our customers goes through, and it updates our billing systesm |
| 13:20.56 | Ice_Strike | So I am guessing you run two VM's for each customer.. Linux Asterisk and Other OS for Dialer System to communicate with asterisk server? |
| 13:21.11 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 13:21.44 | Greenlight | Well, we've another non-virtualised beefy Asterisk box that handles all the "dialler" side of things |
| 13:22.10 | Greenlight | Tbh, we probably could have virtualised it, but at the time we purchsed the kit, I wans't sure |
| 13:22.44 | Ice_Strike | Aha |
| 13:22.46 | Greenlight | And it can sometimes be doing a lot of timing senstive stuff which it's perhaps best to not virtulised |
| 13:23.24 | Ice_Strike | I see your point. |
| 13:23.31 | *** join/#asterisk nbash (46a7e353@gateway/web/freenode/ip.70.167.227.83) |
| 13:25.38 | Ice_Strike | Where are you base? |
| 13:25.41 | nbash | Hello everyone, I have exhausted my searching but can't seem to find a decent example. Is it possible in 11.8.1 to have a dial plan that will redirect 1 specific inbound CCID number to somewhere. EG: If I call from my cell then goto VM of EXT2? |
| 13:25.51 | Greenlight | Dundee, up in bonny Scotland. |
| 13:26.11 | Ice_Strike | I've not been to Scotland for while. |
| 13:28.14 | Greenlight | nbash: exten => 1000/YOURNUMBER,1,Goto() ? |
| 13:28.29 | Greenlight | So calls *to* 1000 *from* YOURNUMBER are picked up |
| 13:28.58 | file | most nights I don't know... anymore woah woah |
| 13:31.00 | nbash | so I could do exten => TRUNKNUMBER/MYCELLNUMBER,1,Goto(9)? |
| 13:31.19 | Katty | hello my asterisk does not work at all how to fix pls |
| 13:31.29 | nbash | then on 9 do whatever |
| 13:31.33 | Greenlight | mhm |
| 13:31.53 | Greenlight | Katty: Sounds like you've powered it up, you'll need to undo that. |
| 13:32.06 | *** join/#asterisk DataWraith (~s1gny@188-194-90-27-dynip.superkabel.de) |
| 13:34.19 | Katty | how to undo powerup?? |
| 13:34.41 | nbash | sorry Katty, I always know I can find answers here...VOIP isn't my specialty however falls into a category I'm being forced to support. I fully understand I need to study more topics on it. Thank you Greenlight for you help |
| 13:34.55 | Greenlight | Scissors |
| 13:35.20 | Greenlight | nbash: I don't think Kattys joke was directed at you |
| 13:35.31 | Katty | it was not. |
| 13:35.43 | Katty | i randomly appear and disappear. |
| 13:35.46 | Greenlight | Yup ^^ |
| 13:35.47 | nbash | lulz |
| 13:35.52 | file | VoIP with Vonage! |
| 13:36.06 | DruidZ | Step 1: Open a window. Step 2: Toss computer out of it. :-) |
| 13:36.29 | Greenlight | If you're on a ground floor, and have a long power cord... |
| 13:38.46 | nbash | well I know it must get old with "n00bs" such as myself randomly popping in here and demanding answers to remedial topics...I honestly did try searching for 15 min with no direct answer...wasn't phrasing the question right in the search context. within 2 mins here I had the answer...again thank you guys, places like this make asterisks community great. |
| 13:38.50 | *** part/#asterisk nbash (46a7e353@gateway/web/freenode/ip.70.167.227.83) |
| 13:39.36 | jameswf | ~book |
| 13:39.36 | 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 |
| 13:39.51 | DruidZ | So, yesterday morning I was running my VoIP users off a FS box. It just stopped working and I couldn't get it going again. This morning those users are running off an Asterisk box. |
| 13:40.15 | DruidZ | I may have a question or two myself. |
| 13:41.02 | DruidZ | However, the online searches and documentation got me up and running so far. :-) |
| 13:41.23 | file | so we don't actually suck with documentation? |
| 13:41.58 | jameswf | Reading is hard.... |
| 13:41.59 | DruidZ | Right now I am looking for some good tutorials on setting up VM. |
| 13:42.23 | DruidZ | The guide was very clear and got me up quite quickly. |
| 13:42.42 | Greenlight | Often some of the conf files themselves have lots of useful info |
| 13:42.53 | Greenlight | (The sample ones) |
| 13:42.56 | file | here's a question... cause it just came to mind and would be sorta cool... |
| 13:43.07 | DruidZ | Yes, I consider those part of the documentation. |
| 13:43.41 | file | what would you (#asterisk) like to know about Asterisk in a blog post from a developer who has been around for 10+ years? |
| 13:43.57 | Katty | eyes file suspiciously |
| 13:44.06 | file | Katty, I'm sourcing blog material! :P |
| 13:44.08 | Katty | hides her blog |
| 13:44.15 | DruidZ | file: Everything. |
| 13:44.21 | jameswf | file: AMA right now... |
| 13:44.44 | jameswf | file: is Mark dreamy |
| 13:44.49 | Katty | oh well if YOU want to source blog material |
| 13:44.53 | Katty | go for it. |
| 13:45.13 | file | jameswf, OFF TOPIC! :p |
| 13:46.13 | bitwize | WIMPy: Sorry for late reply. The channel name does not help me here, I have multiple instances of a subsystem which all connecting to the same AMI manager. I need some logic in the subsystems to ignore events which belongs to another subsystem instance. It is the correct subsystem identifier that are populated in channel variables when channels are created. |
| 13:47.28 | WIMPy | bitwize: Then make yourself a table. |
| 13:49.05 | DruidZ | One of my users just told me that automated services (conferences) won't recognize his touch tones any more since I switched to Asterisk. |
| 13:50.11 | DruidZ | Is default Asterisk swallowing those? How do I pass them through? |
| 13:52.55 | DruidZ | I have dtmfmode=rfc2833 |
| 13:53.18 | DruidZ | I have dtmfmode=rfc2833 set in the connection to my PSTN provider. |
| 13:55.54 | Greenlight | I've had issues with DTMF before; seems some ITSPs disagree with Asterisk on the RFC. |
| 13:56.13 | Greenlight | Specifically Cisco gear uses 101 and asterisk uses 96 (or other way around) |
| 13:56.25 | Greenlight | You can change to inband or just change the source code as we did |
| 13:57.26 | bitwize | DruidZ: I had the same situation with a 3'rd party conference module (app_konference). We control Asterisk via AMI, so we solved it by listening on DTMF events and then passing them further on to related channels. |
| 13:57.51 | bitwize | WIMPy: Thanks for your input. |
| 13:58.15 | Ice_Strike | Some SIP Providers dont work really well with DTMF |
| 13:58.20 | Ice_Strike | I seen this issue before. |
| 13:58.46 | Greenlight | I beleive its the same issue. Granted the RFC is somewhat vague, which doesn't help. |
| 13:58.54 | *** join/#asterisk napnap (~napnap@95-141-110-114.as16211.net) |
| 13:59.43 | Ice_Strike | The DTMF issue I had before that is not accepting key digit. |
| 13:59.53 | Ice_Strike | Changing SIP provider.. work perfect. |
| 13:59.59 | Ice_Strike | Drive me crazy |
| 14:01.10 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:01.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:04.46 | Katty | i finished watching The Borgias. what should i watch now? |
| 14:04.49 | *** join/#asterisk af_ (~af@static-82-85-142-163.clienti.tiscali.it) |
| 14:05.01 | Greenlight | Watched Game of Thrones yet? |
| 14:06.02 | Katty | mmm, the first season i think |
| 14:06.07 | Katty | up to the point where the little dragons hatched |
| 14:06.21 | Katty | was that the first season? |
| 14:06.45 | Greenlight | Maybe into the second. There's lots more to watch and a new season in a couple weeks! |
| 14:07.02 | Katty | should i start all the way over? |
| 14:08.11 | Greenlight | They're just as cool when you watch them again, I reckon so yea, then you'll not be like wondering who or what's happened if you've forgot bits |
| 14:08.19 | Katty | nods |
| 14:08.23 | Katty | have you seen the Tudors? |
| 14:08.52 | Greenlight | Hmm nope, it good ? |
| 14:10.17 | Katty | dunno. it's on netflix and has good reviews tho |
| 14:10.39 | Katty | i'll watch game of thrones tho. |
| 14:10.42 | Katty | it has pretty boys. |
| 14:10.49 | Greenlight | ^^ |
| 14:11.01 | *** join/#asterisk aness (~aness@cm-84.215.76.18.getinternet.no) |
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| 14:11.52 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:12.11 | Greenlight | offline for a bit any moment... engineer just arrived to fix our net connection |
| 14:12.22 | Katty | hopefully it goes quick! |
| 14:16.08 | *** join/#asterisk rox (~rox@89.142.168.6) |
| 14:18.26 | rox | can anybody tell me, how I can control the volume of the whisperer in ChanSpy whisper mode? |
| 14:19.06 | *** join/#asterisk D30 (~deo@112.198.90.25) |
| 14:19.24 | rox | the vX option of ChanSpy seems to only control how loud the whisperer hears the spied channels, is there a way to control, how loudly the spied-on channel hears the whisperer? |
| 14:20.08 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 14:20.38 | napnap | Hi all I try to configure a B400P card..On one hand I've this card (installed on OpenVox with elastix system) on other hand i have two T0 lines...I don't understand the correct way to a good setup...mode TE vs mode NT , BRI vs ISDN :-s |
| 14:22.51 | napnap | First as I understand I would use TE mode, as my T0 lines are from my telecom provider. |
| 14:26.07 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 14:28.34 | bitwize | If anyone was interested in my previous problem I solved it by configuring option "channelvars" in the [general] section of manager.conf, did not work when the option was only configured directly on the manager. |
| 14:28.49 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-aphlnaygrubzfftv) |
| 14:36.14 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
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| 15:07.37 | [TK]D-Fender | [10:19]roxthe vX option of ChanSpy seems to only control how loud the whisperer hears the spied channels, is there a way to control, how loudly the spied-on channel hears the whisperer? <- in the app itself, no... set in the the channel directly via func_volume |
| 15:10.51 | *** join/#asterisk ghost75 (~quassel@ipservice-092-211-045-021.pools.arcor-ip.net) |
| 15:12.37 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
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| 15:19.18 | *** part/#asterisk bulkorok (~Adium@85.183.61.47) |
| 15:27.59 | napnap | anyone can help me to setup my T0 line please ? they are in red alarm and I don't know why (this is my first installation) |
| 15:29.44 | Kobaz | red means no physical connection |
| 15:33.53 | napnap | I will go check the cable but I think the connection is OK... |
| 15:35.47 | jameswf | Be sure you are using an RJ48 crossover and NOT a ethernet crossover.... they are NOT the same. |
| 15:37.40 | MaliutaLap | jameswf: shutup! rj12, rj45, rj48 ... as if my brain wasn't screwed up enough with cable issues ;P |
| 15:38.02 | MaliutaLap | then add shielding ... |
| 15:38.59 | Qwell | MaliutaLap: This will be on the test. |
| 15:40.17 | jameswf | I spent a couple years wiring up very large installs. I find the punch down noise very theriputic.... |
| 15:41.03 | MaliutaLap | Oh, that's not my issue ... making patch cables and punching down into my panel relaxes me. |
| 15:41.30 | MaliutaLap | I just deal with rj45 on about 3 levels (one of my mates does A/V stuff over rj45) |
| 15:41.56 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 15:42.26 | MaliutaLap | then there is rj12 for my POTS things ... adding rj48 to my mind is just getting a bit much. Kinda like how other projects have me writing Python and Perl - and it's rotting my brain |
| 15:48.06 | DruidZ | Python rocks! |
| 15:48.30 | DruidZ | Does Asterisk have a Python interface? |
| 15:49.07 | napnap | Kobaz, jameswf , my cable is right connected. RJ45 or RJ48 no idea, but I use this cable on another asterisk. Is not the same card but I think all T0 card use same sort of wires connections |
| 15:49.10 | DruidZ | And Perl isn't the worst language. |
| 15:49.28 | DruidZ | Of course, it might be if PHP din't exist. |
| 15:49.40 | Qwell | napnap: What is the order of the colors on each end of the cable? |
| 15:49.44 | DruidZ | *didn't* |
| 15:52.20 | napnap | Qwell, mmmh...really you think different T0 cards can use different wires connection ? (I have to walk 1 km to watch the cable :-p ) |
| 15:53.13 | Qwell | well, I would assume that some cards have auto-sensing, like ethernet cards. It's pretty important to have the correct cable on a card that can't. |
| 15:54.09 | MaliutaLap | DruidZ: Py-rub-erl suck ass :P |
| 15:54.47 | MaliutaLap | DruidZ: you could use AGI or other methods to use Python with * - if you were feeling that sick :P |
| 15:55.18 | MaliutaLap | I still prefer old school C |
| 15:55.23 | MaliutaLap | no ++ |
| 15:57.34 | jameswf | DruidZ: here is an example of python with Asterisk https://gist.github.com/jfinstrom/4227521 |
| 16:01.49 | napnap | Qwell, then on ipbx side I can check wire colors, but on onther each wire are connected to phones box (like this) then another cables to an another old(30 years) and big box( on a private area), then are my provider cable. So check cable is not easy for me. |
| 16:02.08 | napnap | liek this => http://www.aduf.org/images/uploads/2861/ed04db4d2dd00dd998ebc13e03f69d42.jpg |
| 16:03.40 | MaliutaLap | napnap: good luck with that cable nest :) |
| 16:04.04 | napnap | :-\ |
| 16:04.08 | MaliutaLap | napnap: all I can recommend is a decent multimeter |
| 16:04.26 | MaliutaLap | tracing anything is going to be a massive PITA |
| 16:04.31 | napnap | I don't have... |
| 16:05.14 | napnap | But, I can perhaps check the model of each T0 cards |
| 16:05.17 | MaliutaLap | investing in a basic digital multimeter and some probes is worth the expense |
| 16:05.30 | napnap | and see if both of they use same wire connection... |
| 16:07.27 | Penguin | You don't even have to buy an $800 Fluke meter. A $40 meter will do just fine for many applications. |
| 16:08.42 | MaliutaLap | Penguin: my mate who does A/V stuff has the most awesome oscilloscope to work with |
| 16:09.12 | MaliutaLap | Penguin: to the point where the engineers that design the shite he fixes want one |
| 16:09.31 | Penguin | I'm about to pick up a used scope. |
| 16:09.39 | Penguin | It's only 60 MHz, though. |
| 16:09.44 | MaliutaLap | Penguin: and i think I paid less than $40AU for my meter |
| 16:10.30 | MaliutaLap | Penguin: I'd have to talk to Rob to get the specs on his kit ... but it integrates with computers (both windows and *nix) for exporting results |
| 16:11.13 | MaliutaLap | I get to play occasionally when I have difficult "issues" - he has the building keys and codes for weekends :) |
| 16:11.24 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 16:12.16 | *** join/#asterisk defswork (~andy@141.0.50.105) |
| 16:12.54 | MaliutaLap | reminds me I need to buy a second set of probes for my meter and solder pins to them so I take readings from batteries and RJ12/45 |
| 16:16.31 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 16:16.32 | DruidZ | jameswf: Thanks. |
| 16:18.49 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 16:19.10 | jameswf | napnap: if you can't see both ends of the cable there is a good chance that is your problem. After 30 years you simply looking at it may have mad it fail. Get a new cable. For giggles you can make an RJ48 loopback by connecting pin 1 -> 4 and 2 -> 5 to see the alarm clear. |
| 16:19.59 | WIMPy | RJ48 is S2M? |
| 16:24.32 | Penguin | Never A2M. |
| 16:24.41 | Penguin | That's how you get pink eye. |
| 16:25.54 | *** join/#asterisk path (~lsanmarti@lsanmartin.uchile.cl) |
| 16:26.00 | path | hey guys |
| 16:26.20 | path | is it possible to modify the FROM field from SIP headers? |
| 16:26.41 | path | thesedays asterisk is doing: "11111111" <sip:11111111111 at 1.1.1.1>;tag=as167b4b82 |
| 16:26.46 | MaliutaLap | Penguin: rofl |
| 16:27.00 | path | I need to replace with this: From :"" <sip:005114824403 at 200.91.0.146;user=tel> |
| 16:27.22 | MaliutaLap | jameswf: I would have gone with "just thinking about the cable made it break" ;) |
| 16:27.53 | path | the idea is replace tag with user=tel |
| 16:28.13 | ghost75 | wtf why my phone is rebooting |
| 16:28.58 | MaliutaLap | ghost75: because you told it to |
| 16:29.15 | ghost75 | by looking at it |
| 16:30.22 | Penguin | Looks can be misinterpreted. |
| 16:30.49 | Penguin | Like that one girl at the Home Depot... |
| 16:30.54 | MaliutaLap | ghost75: these things can read your mind ;) |
| 16:31.10 | ghost75 | so thats a smart phone huh |
| 16:31.42 | MaliutaLap | ghost75: smart phones are all the rage |
| 16:31.57 | ghost75 | all evil |
| 16:32.08 | MaliutaLap | Penguin: you need to find this girl and export her to .au ;) |
| 16:32.15 | WIMPy | What was the? It starts with the wish to conect a T1 and then there comes a crossover? Wheat's that doing there? |
| 16:32.23 | ghost75 | guess psu is dying |
| 16:32.34 | MaliutaLap | ghost75: it's not POE? |
| 16:32.40 | ghost75 | not |
| 16:33.02 | ghost75 | poe is waste of energy |
| 16:33.17 | MaliutaLap | ghost75: and it's not POE capable? so putting a POE injector in front of it is not an issue |
| 16:33.25 | WIMPy | And if anyone can point me to a definition of RJ48, that would be nice. |
| 16:33.27 | *** part/#asterisk bitwize (~S2TS@h87-96-213-2.dynamic.se.alltele.net) |
| 16:33.30 | MaliutaLap | ghost75: POE can be a saviour |
| 16:33.50 | MaliutaLap | WIMPy: google is your friend :) |
| 16:34.13 | WIMPy | Doesn't come up with much info. |
| 16:34.31 | MaliutaLap | WIMPy: I found a wiring outline for it |
| 16:35.02 | Penguin | https://supportforums.cisco.com/discussion/9134546/rj48c-vs-rj45 |
| 16:35.23 | MaliutaLap | WIMPy: http://www.juniper.net/techpubs/en_US/release-independent/junos/topics/reference/specifications/rj48-to-rj48-crossover-connector-pinouts-t1e1-mini-pim.html http://www.ehow.com/info_8349732_differences-rj45-rj48.html |
| 16:35.23 | WIMPy | I see wirings for some devices, but not what interface it is. |
| 16:36.31 | MaliutaLap | WIMPy: https://en.wikipedia.org/wiki/Registered_jack#RJ48 |
| 16:36.40 | MaliutaLap | WIMPy: enough info for you? |
| 16:36.59 | *** join/#asterisk navaismo (~navaismo@201.124.129.180) |
| 16:37.09 | ghost75 | as i remember crisco has its own poe standard |
| 16:37.11 | WIMPy | No mention of the kind of interface. It could be S2, but I don't know what would be used for U2 in T-areas, so it cuold be that as well. |
| 16:38.38 | Penguin | There's a legacy Cisco PoE, but even the legacy devices often supported 802.3af still. |
| 16:38.59 | WIMPy | "Inline Power" |
| 16:39.26 | Penguin | The Cisco legacy devices would accept power on the unused pairs. |
| 16:39.32 | MaliutaLap | ghost75: what's the phone? you can get injectors to work with most cisco devices |
| 16:39.49 | ghost75 | 7941 |
| 16:40.18 | MaliutaLap | ghost75: I have 7942 and I can get injectors for less that $10US off of ebat |
| 16:40.25 | MaliutaLap | s/ebat/ebay/ |
| 16:40.43 | Penguin | I just build my own with a DC power jack and a couple Cat 5e jacks. |
| 16:41.19 | ghost75 | as such ? http://www.wifi-stock.com/prods/POE-INJ.jpg |
| 16:41.53 | Penguin | Mine isn't as pretty, but that's the same thing. |
| 16:42.13 | ghost75 | somewhere i have one laying i think |
| 16:43.02 | DruidZ | "No application 'VoicemailMain'" do I need to load a module or something? All the tutorials seem to assume that the command just works. |
| 16:43.20 | MaliutaLap | ghost75: if it's the "powercube" and it's jack port that are dying it might be worth putting the POE injector inline |
| 16:43.44 | ghost75 | but then still need a psu |
| 16:43.55 | MaliutaLap | DruidZ: is voicemail configured? is the voicemail module included? |
| 16:43.59 | *** join/#asterisk vlad_sta_ (~vlad_star@nat.canmos.ru) |
| 16:44.53 | MaliutaLap | ghost75: from my understanding of the 79xx hardware the POE is separate to the "powercube" input |
| 16:45.03 | MaliutaLap | ghost75: but I may be wrong about that |
| 16:45.42 | ghost75 | i got generic 48v psu |
| 16:46.03 | DruidZ | MaliutaLap: I can't find a voicemail module in the sample modules file. |
| 16:46.11 | ghost75 | also i think the poe injectors is doing only up to 12v |
| 16:46.39 | Penguin | When I built the inline power for my 7900 series phones, I removed the power cubes from the jacks on the phones and connected them to the DC jacks on the inline power injectors I built. |
| 16:46.43 | MaliutaLap | ghost75: you're not using a cisco PSU with that phone? |
| 16:46.54 | ghost75 | nope |
| 16:47.04 | MaliutaLap | steps back |
| 16:47.13 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 16:47.16 | MaliutaLap | I wouldn't have put a generic in the loop |
| 16:47.28 | Penguin | You need the 48V on the ethernet jack just the same as you need it on the DC jack. |
| 16:47.35 | MaliutaLap | DruidZ: I don't have my * box up at the moment, can't check anything for you |
| 16:48.12 | Penguin | druidz: load app_voicemail.so |
| 16:48.16 | MaliutaLap | DruidZ: and you haven't mentioned what version of * you're running |
| 16:48.27 | MaliutaLap | although what Penguin said |
| 16:48.34 | Penguin | If you have autoload enabled and have a good voicemail.conf, the module should have loaded automatically. |
| 16:49.02 | MaliutaLap | autoload? and get all the crap you don't need bloating the system ;_ |
| 16:49.10 | MaliutaLap | s/:_/;)/ |
| 16:49.19 | ghost75 | TP-Link Passive PoE injector <- why this has voltage limit |
| 16:49.24 | Penguin | Yes, autoload. If you don't have confs for something, it doesn't load. |
| 16:49.33 | MaliutaLap | TP-Link - now that's a problem |
| 16:49.45 | Penguin | Autoload basically means if there is a good conf for it, load the module because you probably want to use it. |
| 16:49.48 | MaliutaLap | wouldn't touch TP-Link with someone elses 10 foot pole |
| 16:50.01 | rrittgarn | how can a PoE injector be passive? |
| 16:50.04 | ghost75 | tp link is great for soho |
| 16:50.13 | ghost75 | http://www.olivershingler.co.uk/oliblog/wp-content/uploads/2012/12/TPLINK-WAP-POE.jpg |
| 16:50.18 | DruidZ | MaliutaLap: Got it. I also have to load res_smdi I guess. |
| 16:50.22 | MaliutaLap | ghost75: no, linsys or cisco are good for SOHO |
| 16:50.22 | DruidZ | Version 1.6 |
| 16:50.28 | DruidZ | Version 1.8 I mean. |
| 16:50.38 | ghost75 | but no good for * :< |
| 16:50.57 | WIMPy | DruidZ: Why didn't you use 11? |
| 16:51.01 | MaliutaLap | ghost75: that's a separate debate - I don't mind Cisco phones with SIP |
| 16:51.09 | Penguin | druidz: (1148.34) <Penguin> If you have autoload enabled and have a good voicemail.conf, the module should have loaded automatically. |
| 16:51.23 | ghost75 | and since linksys went to belkin, i dont like it anymore |
| 16:51.45 | Penguin | What means "linksys went to belkin" ? |
| 16:51.48 | MaliutaLap | ghost75: yeah, cisco did pull all the good bits out of linksys before selling it |
| 16:51.55 | Penguin | Cisco sold it? |
| 16:52.01 | Penguin | I had not heard about this! |
| 16:52.03 | DruidZ | I am running NetBSD and it hasn't been ported yet. |
| 16:52.18 | [TK]D-Fender | [12:50]ghost75http://www.olivershingler.co.uk/oliblog/wp-content/uploads/2012/12/TPLINK-WAP-POE.jpg <- that isn't on their site, and I see no confirmation of what it puts out. Their other stuff is 802.3af, which most older 7900 series don't support |
| 16:52.19 | MaliutaLap | Penguin: you didn't know about the Cisco bought Linksys then sold it thing? ... have you been living in cave? |
| 16:52.38 | MaliutaLap | Penguin: that happened ages ago |
| 16:52.44 | Penguin | I knew Cisco took Linksys some time ago, but I never heard anything about Belkin. |
| 16:53.12 | MaliutaLap | Penguin: http://www.bloomberg.com/news/2013-01-24/cisco-sells-linksys-home-router-unit-to-belkin.html |
| 16:53.25 | ghost75 | they wanted to keep silent about it i guess |
| 16:53.52 | MaliutaLap | ghost75: yeah right, they tendered it through Loyds - it didn't stay quiet for long |
| 16:55.14 | MaliutaLap | Penguin: they stripped a whole heap of SOHO stuff from the Linksys VoIP and networking ranges before offloading it |
| 16:55.40 | ghost75 | what about linksys switches |
| 16:56.22 | MaliutaLap | ghost75: read "networking" |
| 16:56.50 | MaliutaLap | ghost75: I have a cisco SLM here, that has been superceded by stuff wrenched from Linsys |
| 16:57.07 | ghost75 | 2y ago i purchased linksys switch that looked like it was produced by cisco |
| 16:57.12 | DruidZ | exten => *98,1,VoicemailMain(s${CALLERID(num)}) |
| 16:57.32 | MaliutaLap | ghost75: 2 years ago Cisco owned Linksys |
| 16:57.34 | DruidZ | That still asks for a login and password. |
| 16:57.34 | [TK]D-Fender | DruidZ: not valid. |
| 16:57.43 | [TK]D-Fender | duino "s" infront. This isn't * 1.0 |
| 16:57.53 | [TK]D-Fender | DruidZ: no "s" infront. This isn't * 1.0 |
| 16:58.04 | WIMPy | DruidZ: That syntax was was valid 6 years ago. |
| 16:58.19 | DruidZ | Guess it was an old web site. |
| 16:58.25 | [TK]D-Fender | actually... earlier |
| 16:58.43 | [TK]D-Fender | DruidZ: "core show application XYZ" |
| 16:58.45 | MaliutaLap | [TK]D-Fender: stop killing the buzz man ;{ |
| 16:58.49 | WIMPy | DruidZ: 'core show application VoiceMailMain' |
| 16:59.06 | MaliutaLap | should go to bed |
| 16:59.24 | MaliutaLap | not as drunk as I was when I got home from the gig |
| 16:59.56 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 17:02.10 | DruidZ | exten => *98,1,VoicemailMain(${CALLERID(num)},s) still asks for a password. |
| 17:02.26 | [TK]D-Fender | Show us... |
| 17:02.46 | [TK]D-Fender | ~pb |
| 17:02.46 | 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. |
| 17:02.48 | MaliutaLap | [TK]D-Fender: don't you mean "show me the money" ;) |
| 17:02.48 | [TK]D-Fender | ^^^^^ |
| 17:05.24 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 17:06.33 | MaliutaLap | really goes to bed |
| 17:10.39 | DruidZ | Sorry for the noobiness. I had to learn and install Asterisk in one day. |
| 17:10.50 | DruidZ | isn't getting much sleep these days. |
| 17:11.28 | jameswf | ~noob |
| 17:11.39 | jameswf | sigh |
| 17:12.54 | WIMPy | DruidZ: Next time you install something new, use soem current version. |
| 17:13.04 | DruidZ | Feel free to hi tme with the Clue-By-Four. It'll probably wake me up. |
| 17:13.26 | DruidZ | I hope to port a newer version over as soon as I have this working. |
| 17:13.38 | DruidZ | is also a NetBSD developer. |
| 17:13.47 | WIMPy | I prefer poster rolls filled with concrete :-) |
| 17:15.03 | *** join/#asterisk jmls (~julian@77.107.171.82) |
| 17:15.13 | jmls | good afternoon all |
| 17:15.25 | jmls | been playing with some of the new ami events in 12 |
| 17:15.32 | DruidZ | Actually version 11.7 is ported but is has some issues that need to be addressed. |
| 17:15.46 | jmls | if 502 dials 503, get the expected dialBegin and dialEnd ami events |
| 17:16.33 | jmls | but if I dial (say) an extension that answers and plays moh, there are no BeginDial and DialEnd events |
| 17:16.57 | jmls | why is this ? is it because there isn't a destination channel ? |
| 17:17.16 | WIMPy | You expect Dial events if you don't dial? |
| 17:18.04 | jmls | WIMPy: it's nomenclature. I "dialed" 1000 on the keypad, and pressed the "dial" button |
| 17:18.27 | jmls | I know that the Dial() application wasn't called |
| 17:18.28 | WIMPy | That's not what it's about. |
| 17:18.38 | jmls | yeah. I figured that ;) |
| 17:18.45 | WIMPy | That however is. |
| 17:19.04 | WIMPy | And if you press a send button, that's not dialling, either. |
| 17:19.45 | WIMPy | there is no communication with Asterisk before you hit the send key. |
| 17:20.08 | jmls | I was just hoping to use the Dialx events to track the state of a device. So I now have to use 2 different events (one for dialling a channel, and one for just calling into the dialplan). Either way, the device is "busy" |
| 17:20.47 | WIMPy | Theres even more: It could receive a call. |
| 17:21.00 | jmls | aw crap. |
| 17:21.12 | jmls | cel it is then |
| 17:21.28 | WIMPy | But in all cases you get the same newstate event. |
| 17:21.55 | jmls | i'm surprised that there isn't a "device" event |
| 17:22.25 | jmls | but nevermind |
| 17:22.35 | jmls | let me go back to my tinkering . thanks. |
| 17:22.59 | WIMPy | There is an extensionstatus event if you have a hint. |
| 17:25.13 | jmls | thanks |
| 17:25.51 | jmls | I was playing with the newstate event, but was slightly confused: take the same scenario, 502 calling 1000 to play moh |
| 17:26.17 | jmls | I get an event a) channelstatedesc: 'Ring' |
| 17:26.23 | jmls | and b) channelstatedesc: 'Up' |
| 17:26.39 | jmls | but no newstate for hangup. |
| 17:26.47 | jmls | There is a hanguprequest event |
| 17:27.09 | WIMPy | And a hangup event. |
| 17:27.35 | WIMPy | That's the other end of the newchannel event. |
| 17:28.26 | jmls | again, I just thought that there would be a newstate event for the hangup. It's not a problem, just feels inconsistent |
| 17:28.42 | jmls | hannelstatedesc: 'Down' |
| 17:28.47 | jmls | or something like that . |
| 17:29.01 | WIMPy | That's because it's not a new state of the channel. It's the channel being destroyed. |
| 17:29.12 | *** join/#asterisk calum_ (~calum_@92.40.249.209.threembb.co.uk) |
| 17:29.28 | jmls | fair enough. I surrender ;) |
| 17:29.54 | WIMPy | \o/ |
| 17:30.40 | jmls | am I right in saying that the extensionstatus event only happens on channels that a) have a hint, and b) that hint has a subscription ? |
| 17:30.43 | ghost75 | when i send fax and it says faxopt(status) success and faxopt(error) hangup, was the fax sent or not? |
| 17:31.25 | WIMPy | It doesn't happen on the channel, it happens on the hints. So yes for that part. But you don;t have to have an active subscription. |
| 17:32.33 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 17:32.36 | jmls | WIMPy: I have a "generic" hints |
| 17:32.54 | jameswf | ghost75: Schrodinger's fax? |
| 17:32.59 | jmls | exten => _[0-9A-Za-z*+].,hint,${DB(somefamily)} |
| 17:33.25 | jmls | I have 2 phones, 502 and 503. 503 subsribes to 502 |
| 17:33.29 | ghost75 | fax show stats thinks it was successfull |
| 17:33.58 | jmls | 503 "sees" the hints, and does things with it's blinky lights, so I know that it works, and the hint has 1 watcher |
| 17:34.25 | jmls | but extensionstatus event is only fired when 502 does something |
| 17:34.34 | jmls | if 503 does something, the event is not fired |
| 17:35.01 | jmls | so that's why I assumed that there had to be an active subscription. |
| 17:36.14 | WIMPy | Maybe there's a difference when using patterns. I don't know. |
| 17:36.25 | WIMPy | But with static hints you don't need watchers. |
| 17:38.46 | *** join/#asterisk brittel (~jack@542713e4.ftth.concepts.nl) |
| 17:40.53 | ghost75 | so maybe faxopt(error) was still hangup because there was no error in last fax ? |
| 17:44.12 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 17:51.49 | DruidZ | Yay! Voice mail is working. |
| 17:53.15 | Katty | VMWARE PEN. |
| 17:57.54 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:59.24 | ghost75 | jameswf: the cat was prolly alive |
| 18:02.20 | ghost75 | * is like schrodingers box |
| 18:07.11 | file | looks around |
| 18:13.21 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:13.41 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 18:15.25 | *** join/#asterisk jpoz_ (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 18:20.48 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 18:21.24 | *** join/#asterisk serafie1 (~erin@nat/digium/x-ghejsexvjhkociku) |
| 18:25.13 | *** join/#asterisk Devon_ (~Devon_@63.214.236.169) |
| 18:27.02 | *** join/#asterisk yoavz (yoavz@yoavz.net) |
| 18:28.57 | *** join/#asterisk serafie (~erin@nat/digium/x-hfwagvfzdwjsnaxy) |
| 18:31.14 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:31.28 | DruidZ | Is there anyone around that can call a SIP address? |
| 18:31.34 | DruidZ | For a test. |
| 18:47.00 | *** join/#asterisk illuder (illuder@105-236-34-179.access.mtnbusiness.co.za) |
| 18:50.36 | Penguin | What's the URI? |
| 18:51.06 | Penguin | druidz: URI? |
| 19:00.32 | DruidZ | Penguin: Did you get my /msg? |
| 19:01.11 | DruidZ | In fact, if anyone wants to call me it's sip:darcy@vex.net |
| 19:01.16 | Penguin | Nope, privmsg is blocked until I know someone is going to be sending me something in private. |
| 19:08.02 | DruidZ | Did anyone try it? Please let me know the error if you did and it failed. |
| 19:13.36 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 19:16.48 | Penguin | I sent a call to you, but it failed with a warning about retransmission timeout being reached. I often see that when there is a NAT problem. |
| 19:17.46 | *** join/#asterisk serafie (~erin@nat/digium/x-btlvdlpelyzbsfir) |
| 19:20.54 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 19:21.33 | *** join/#asterisk xinit (wouter@ipv6.pruts.nl) |
| 19:22.40 | *** join/#asterisk spicyramen_ (~spicyrame@c-98-210-160-181.hsd1.ca.comcast.net) |
| 19:23.16 | xinit | hi! I'm trying to setup dialing out over SIP to twilio and I would like to play an announcement to the callee (A(mymesg) option in Dial()), but for some reason, asterisk thinks that the call has already been answered by twilio, while the phone is still ringing. Is there something I can do to wait until the phone of the callee is really picked up? |
| 19:24.01 | Penguin | Did you check the SIP debug to see where/when/why the call is being answered? |
| 19:24.16 | Penguin | It could also be related to your dialplan. |
| 19:26.34 | DruidZ | Penguin: The switch is not behind a firewall. |
| 19:26.36 | xinit | Penguin: yes, or at least, I'm running an asterisk -rvvv to see what's going on |
| 19:27.26 | xinit | Penguin: it says SIP/twilio-00000037 answered SIP/myclient-00000036 |
| 19:27.40 | Penguin | xinit: That in itself doesn't show sip debugging. You must set sip debug to enabled to see that info. |
| 19:28.02 | xinit | Penguin: ah, cool, let me check |
| 19:28.11 | Penguin | sip set debug on |
| 19:28.30 | Penguin | I don't know anything about twilio, so I don't know what might make it answer right away. |
| 19:28.46 | spicyramen_ | hi x init sip debug will speak by itself or packet capture |
| 19:29.09 | spicyramen_ | tcpdump |
| 19:29.36 | Penguin | But if it plays sounds and has silence after being answered, asterisk does have a couple applications that will detect that behavior and you can wait for certain criteria before playback. |
| 19:29.44 | spicyramen_ | I know twilio r u doing a direct sip trunk ? |
| 19:31.56 | xinit | Penguin: ok.. it seems like it's silent and then plays a ring on the twilo side.. btw, what should I be looking for regarding "where/when/why the call is being answered"? |
| 19:33.22 | spicyramen_ | u can probably see twilio debug tools |
| 19:33.24 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 19:33.29 | Penguin | I'm not entirely sure. You should send an INVITE to the peer and it should respond. The response could be helpful. |
| 19:33.38 | spicyramen_ | At sip level probably they just a sip 200 ok |
| 19:33.48 | spicyramen_ | with sdp |
| 19:34.00 | spicyramen_ | Let us know what u find in sip trace |
| 19:34.01 | Penguin | That probably isn't useful, then. |
| 19:34.20 | spicyramen_ | It could be your twilio config |
| 19:34.28 | spicyramen_ | What do u have configured ? |
| 19:34.37 | spicyramen_ | Can u configure |
| 19:34.54 | xinit | not much you can configure there |
| 19:35.14 | xinit | let me get pastebin the trace.. hold on |
| 19:35.29 | Penguin | Is the twilio something the user has to configure himself or do they just provide a one-size-fits-all deployment? |
| 19:35.45 | spicyramen_ | They allow u to configure urls |
| 19:35.55 | spicyramen_ | Where u launch it php |
| 19:36.02 | spicyramen_ | or other code |
| 19:36.37 | Penguin | I guess it must be the code that is answering the call, then. |
| 19:38.31 | xinit | http://pastebin.com/vi78ayk1 |
| 19:40.06 | [TK]D-Fender | it answered |
| 19:40.16 | [TK]D-Fender | we saw that before asking for SIP debug... |
| 19:40.19 | [TK]D-Fender | we knew it was there... |
| 19:40.25 | [TK]D-Fender | but now you can see it in full |
| 19:40.38 | *** join/#asterisk Rahail (~Rahail@67.214.121.181) |
| 19:41.01 | Rahail | Hi there I need little help with creating Group caller id on a out going trunk or sip user |
| 19:41.35 | [TK]D-Fender | No such thing as a "group caller id" |
| 19:41.37 | Rahail | We are doing auto call to brodcast msg and its showing only 1 caller id number on outgoing is there way we can put group of numbers and it use from there |
| 19:41.41 | [TK]D-Fender | Just set the callerID. Done. |
| 19:42.18 | Penguin | Each outgoing call can have the callerID set. What more do you want? |
| 19:42.18 | [TK]D-Fender | Your description is probably missing a lot of detail... |
| 19:42.24 | Rahail | each extension getting 1 dedicated number but when we doing group call it only show i caller id :( and that number all people calling so If we could add group of callr id.. |
| 19:42.32 | Rahail | its msg brodcasting we trying to do |
| 19:42.44 | [TK]D-Fender | What is a "group call"? |
| 19:42.49 | [TK]D-Fender | What are you expecting to see? |
| 19:42.55 | Rahail | letsay We have 100 Customer and we send call to them |
| 19:43.02 | Penguin | Sounds like a caller ID number rotation would be in order. |
| 19:43.12 | Rahail | something like that how can I add this on trunk |
| 19:43.27 | [TK]D-Fender | There is no such thing as "add this on trunk". |
| 19:43.29 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 19:43.32 | Penguin | It has nothing to do with a "trunk" and everything to do with your dial plan. |
| 19:43.35 | [TK]D-Fender | TYour dialplan does whatever you tell it to.] |
| 19:43.42 | [TK]D-Fender | Your dialplan does whatever you tell it to. |
| 19:43.56 | Rahail | any Idea how can I add this on dailplan to use like 15 Caller id |
| 19:44.03 | [TK]D-Fender | Use HOW? |
| 19:44.04 | [TK]D-Fender | When? |
| 19:44.07 | [TK]D-Fender | based on what decision? |
| 19:44.12 | [TK]D-Fender | How are these calls being placed? |
| 19:44.19 | Rahail | we are ussing SIPP |
| 19:44.21 | Rahail | to call |
| 19:44.21 | [TK]D-Fender | You are being VERY vague on the details |
| 19:44.35 | xinit | Penguin: you said that there are applications that can detect silence and stuff before moving on. Which ones are that? |
| 19:44.36 | Rahail | and under sipp call list we put our customer number's |
| 19:44.49 | Penguin | I would create a list of phone numbers to use as a callerid number. I would create the list in the asterisk DB and assign each one an index number. |
| 19:45.23 | Penguin | xinit: There's WaitForSilence and there's BackGroundDetect. |
| 19:45.37 | Penguin | and WaitForNoise |
| 19:45.39 | xinit | Penguin: thanks |
| 19:45.49 | [TK]D-Fender | "WaitForSilence" IIRC |
| 19:46.10 | Rahail | how will I rotate them.. so 15(did) we have so we can put those and it rotate and each custoemr get diffrent number.. that way they all dont call in same number for inoformaion @ [TK]D-Fender and Penguin |
| 19:46.14 | Penguin | Depends on if you are waiting for noise or waiting for silence. |
| 19:46.30 | [TK]D-Fender | Rahail: How do YOU want them handed out? |
| 19:46.47 | Rahail | random |
| 19:46.50 | [TK]D-Fender | Rahail: YOU have to program the method you want to use to determine this. |
| 19:47.06 | [TK]D-Fender | "core show function like RAND" <------- |
| 19:47.20 | Rahail | is it possible to add this on outgoing exten so that way it rotate those number |
| 19:47.24 | Penguin | Based on the info you have provided so far, I would create a bit of dialplan to set the the callerID number based on the index and increase the index with each call. |
| 19:47.40 | Penguin | It's not random, but it is changing each time. |
| 19:47.43 | *** join/#asterisk scouture (~scouture@unaffiliated/scouture) |
| 19:47.49 | Rahail | yes Penguin that will work |
| 19:47.59 | Rahail | how can i achive it from dialplan |
| 19:48.10 | Penguin | You could also create the same indexing and use the RAND function to randomly pick from the list. |
| 19:48.10 | [TK]D-Fender | "core show function like RAND" <------- |
| 19:48.42 | [TK]D-Fender | Pick random number. use as index to pull some value you put in AstDB. Or use numbered constants |
| 19:48.43 | [TK]D-Fender | Or whatever |
| 19:48.50 | Penguin | It might even be a couple less lines of dialplan to RANDomize it rather than increment it with each call. |
| 19:49.29 | Rahail | Penguin do you think you can give me those lines so I can add it to my dialplan |
| 19:49.58 | Penguin | Not only do you want the ideas, but you want me to do the work for you as well? |
| 19:50.19 | Rahail | I have no idea how to do it this why |
| 19:50.22 | [TK]D-Fender | And assume your patterns or where to put it in the first place? |
| 19:50.25 | [TK]D-Fender | ~book |
| 19:50.26 | 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 |
| 19:50.46 | [TK]D-Fender | ^^^^^^^ |
| 19:50.59 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 19:51.20 | dan_j | Hi, in Asterisk 11, is there any way to reload sip.conf without losing all the realtime sip registrations? |
| 19:53.00 | *** join/#asterisk vlad_sta_ (~vlad_star@77.41.82.191) |
| 19:53.01 | skrusty | not that i know of |
| 19:53.21 | Rahail | no clue lots of thing to go over |
| 19:53.53 | [TK]D-Fender | No time like the present |
| 19:53.59 | [TK]D-Fender | Hit the books... |
| 19:54.01 | Penguin | Set(CALLERID(num)=${DB(callerid/${RAND(0,14)}/list)}); |
| 19:54.21 | Penguin | You'll have to figure out what to do with it. |
| 19:54.38 | *** join/#asterisk Milarepa (~Milarepa@67-2-157-80.slkc.qwest.net) |
| 19:55.02 | Rahail | insted of DB can this pull from some sort of .txt file rest i can try cange around and see if i can make ti work |
| 19:55.22 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:55.30 | Penguin | Sure, but I'm not going to write the code to do that. |
| 19:56.06 | Penguin | I like to do this sort of thing using the asterisk DB, so that's how I would do it. |
| 19:56.21 | Rahail | Please if you can would be grate .. I am stuck in this.. and every one calling same number where other agent setting idel |
| 19:56.51 | Penguin | I would populate 15 keys, each with an incrementing index number, and each with a different phone number to be used as the outgoing callerid number. |
| 19:57.31 | Penguin | I would then set the callerid num from the db keys, either incrementing or randomized. Random seems like less to maintain. |
| 19:57.58 | Penguin | If you increment, then you have to keep track of the index number last used and increment it with each call. That adds more lines of dialplan. |
| 19:58.08 | [TK]D-Fender | Rahail: You aren't "stuck" That would imply haven't actually moved in the direction you wanted to go. Here you're asking us to program your system for you based on vague descriptions with no idea how it needs to be set up for the way your system is laid out. |
| 19:58.22 | Penguin | The random way makes better sense to me for someone who isn't willing to write his own dialplan. |
| 19:58.36 | [TK]D-Fender | Rahail: Hire a consultant |
| 19:58.50 | Penguin | We've already done most of the work! |
| 19:59.01 | Katty | puts a hat on Penguin |
| 19:59.09 | Penguin | Oooh, soft. |
| 19:59.44 | Rahail | Penguin thank you so much.. atlist i got how to start it let see if i can figerout it out |
| 19:59.46 | Rahail | thank you again |
| 19:59.47 | Katty | what if it's a spiky hat? |
| 19:59.59 | Penguin | Oooh, pointy. |
| 20:00.04 | Katty | rolls eyes |
| 20:00.28 | Katty | infobot: Penguin |
| 20:00.28 | infobot | it has been said that penguin is the mascot for Linux. or a fabulous bzflag player |
| 20:00.44 | Katty | puts a fabulous bzflag hat on Penguin |
| 20:01.36 | rrittgarn | dan_j: if you have an individual peer you need to update, you can sip prune realtime peer [Peername] to pull that one out of memory until it registers again. helpful for individual changes |
| 20:04.37 | MaliutaLap | squishes Kattys squirrels ;) |
| 20:06.58 | Katty | hello there mister MaliutaLap! |
| 20:07.04 | Katty | how're you this lovely afternoon? |
| 20:07.31 | MaliutaLap | fine on this at 06:00 this morning. Went to a great gig last night with my uncle |
| 20:07.57 | MaliutaLap | day being slightly thrown off by having to package python crap and write perl scripts |
| 20:08.09 | MaliutaLap | both of which rot the brain |
| 20:11.54 | MaliutaLap | Katty: and how is life in the southern US? |
| 20:12.56 | Katty | nods |
| 20:13.14 | Katty | both certainly do have a way of wanting to pull out portions of your brain out the ear, using a fork |
| 20:13.18 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 20:13.37 | Katty | MaliutaLap: hopefully your day will improve and be wonderful! |
| 20:13.40 | sekil | is there any way to make local channel inherit vars from some sip user? |
| 20:14.27 | Katty | MaliutaLap: so far, mine has been quite lovely and with any luck the weather will hold and i can go jogging this afternoon! |
| 20:14.54 | Katty | now if only Nugget were closer so i could jog with him. |
| 20:15.30 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 20:15.43 | Nugget | I signed up for the RR100 next year. |
| 20:15.56 | Nugget | not sure if that's awesome or the worst idea ever |
| 20:16.15 | MaliutaLap | I'd go jogging ... but the peripheral neuropathy makes the pain in my feet too much - oh and the pred dosage means I bleed muscle mass :( |
| 20:16.48 | MaliutaLap | I blame my oncologist[s]. |
| 20:17.23 | MaliutaLap | on the plus side I got a veiled mention in .au parliament this week |
| 20:19.18 | Katty | MaliutaLap: i can't say i really understand all of what that means, but i'm sorry )= it can't be pleasant |
| 20:19.25 | Katty | Nugget: i think it's a wonderful idea! |
| 20:19.31 | Katty | Nugget: you'll do great!!! |
| 20:19.46 | Nugget | I appreciate your confidence :) |
| 20:20.36 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 20:20.40 | MaliutaLap | Katty: it's simples - peripheral refers to the peripheral parts of the body (feet, fingers). Neuropathy is nerve death (and nerves grow back at a rate of 6mm/month) |
| 20:21.05 | MaliutaLap | Katty: most medical terms are relatively easy to break down and understand :) |
| 20:22.02 | MaliutaLap | and the parliamentary thing is like if your congress person mentioned you in congress :) |
| 20:23.41 | Nugget | has been to canberra |
| 20:24.01 | *** join/#asterisk devlap (~Developme@static-100-32-93-142.lsanca.fios.verizon.net) |
| 20:24.24 | [TK]D-Fender | loves cranberries |
| 20:24.33 | MaliutaLap | Nugget: but did you get an un-escorted pass to parliament house??? And I don't mean the public parts :) |
| 20:24.41 | MaliutaLap | [TK]D-Fender: they are a good band |
| 20:24.44 | devlap | Can I set a per sip.conf context externaddr? |
| 20:24.52 | Penguin | no |
| 20:24.54 | Katty | MaliutaLap: ah right. |
| 20:24.58 | Katty | MaliutaLap: that does make more sense. |
| 20:24.59 | [TK]D-Fender | MaliutaLap: O'Riordan != O'Riginal ;) |
| 20:25.01 | devlap | How would I do that then? |
| 20:25.17 | devlap | On one sip.conf I want to use one ip in the headers, in the other, a different ip. |
| 20:25.17 | [TK]D-Fender | devlap: You can't |
| 20:25.17 | Katty | [TK]D-Fender: i predict you like vodka with your cranberries |
| 20:25.23 | Penguin | Asterisk only has one externaddr. Set it accordingly. |
| 20:25.36 | [TK]D-Fender | devlap: Perhaps PJSIP allows something like this in 12+\ |
| 20:25.37 | Nugget | I can't imagine the private parts are any more interesting than the public chambers :) |
| 20:25.38 | MaliutaLap | devlap: try separate * instances :) |
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| 20:25.43 | [TK]D-Fender | devlap: chan_sip does not |
| 20:25.53 | MaliutaLap | Nugget: you'd be wrong there |
| 20:26.00 | Nugget | https://macnugget.org/albums/australia2002/IMG_1774.jpg |
| 20:26.03 | Nugget | nutty oz |
| 20:26.19 | devlap | So if I have two sip providers, I can't tell asterisk to use different "contact me here" pubic ips ? |
| 20:26.19 | MaliutaLap | Nugget: but then again I am an active member of the ALP :) |
| 20:26.36 | [TK]D-Fender | Alrighty... checkout time... BBIAB |
| 20:26.51 | [TK]D-Fender | devlap: Not with Chan_sip |
| 20:27.00 | MaliutaLap | Nugget: you took a photo of metal representations of food? ;) |
| 20:27.05 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:27.32 | devlap | Well thats a pain in the ass. I have two dedicated links to two different providers at this location. |
| 20:27.33 | Nugget | https://macnugget.org/albums/australia2002/IMG_1765.jpg <-- food |
| 20:27.35 | *** join/#asterisk petris (~petris@192.184.93.147) |
| 20:28.03 | MaliutaLap | Nugget: no, crappy piss |
| 20:28.18 | MaliutaLap | food is what we put on the coat of arms :P |
| 20:28.22 | Nugget | heh |
| 20:28.31 | Katty | that does not look like tasty food to me :< |
| 20:28.37 | Katty | but that DOES look like a very nice scarf. |
| 20:28.45 | MaliutaLap | Katty: Roo and Emu are awesome foods |
| 20:29.00 | MaliutaLap | Katty: the VB is just wrong |
| 20:29.47 | devlap | I have one public range we use for most everything. I got a sip provider just online to use. We recently got a local sip provider as well. They installed a wireless link on the roof, and a router. We basically get dhcp from it, and send some sip out. So if I don't set externaddr am I fine? what could I do in this situation? |
| 20:30.34 | Penguin | If it's local, your externaddr is unimportant. |
| 20:32.04 | devlap | If I try to register a sip session, won't they send back requests to the externaddr which is put in the header? |
| 20:32.06 | Penguin | externaddr is for NAT purposes. |
| 20:32.28 | Katty | i'll nat your purpose in a minute. |
| 20:32.38 | Penguin | Is that a promise? |
| 20:32.42 | devlap | Both connections utilize nat. |
| 20:32.52 | Katty | debates. |
| 20:33.07 | Katty | Penguin: will there be cookies? |
| 20:33.14 | Penguin | You can count on it. |
| 20:33.34 | Katty | woot! |
| 20:33.39 | devlap | our local subnet is 10.42.0.0/21 which asterisk is a part of. the router has dhcp to their routerboard on 10.149.0.0 which im fairly certain is a nat'd "lan" |
| 20:36.14 | MaliutaLap | stalks Katty via the Penguin based cookies ;) |
| 20:39.32 | MaliutaLap | it would appear that Nugget is in what is now the "sister state" to where I live ... but my state is bigger than his ;) |
| 20:43.35 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 20:46.22 | Katty | MaliutaLap: no stalking required! |
| 20:46.22 | *** join/#asterisk fabiobik (~fabio@a89-154-223-3.cpe.netcabo.pt) |
| 20:46.28 | fabiobik | Hello guys |
| 20:46.31 | brittel | Anyone has any experience with SimplOPAL? |
| 20:46.40 | Katty | glances at fabiobik |
| 20:46.51 | Katty | coughs a bit |
| 20:46.58 | MaliutaLap | Katty: are you throwing yourself at me ... I tend not to pick up on these things until someone is naked :) |
| 20:46.58 | fabiobik | How to send sms instead of caller id be a number, send as text? |
| 20:47.12 | *** join/#asterisk vlad_sta_ (~vlad_star@77.41.82.191) |
| 20:47.25 | Katty | MaliutaLap: if i happen to throw myself, it would be greatly appreciated if you don't let me fall face first into the concrete. |
| 20:49.48 | MaliutaLap | Katty: hmm, pretty hard to do from across the Pacific ;) |
| 20:49.53 | MaliutaLap | Katty: I can try though |
| 20:51.42 | fabiobik | how to send a business name or custom text as my caller ID? |
| 20:52.08 | Penguin | Set(CALLERID(name)=Business Name Here) |
| 20:52.15 | rrittgarn | talk to your ITSP 99% of the time what you send gets ignored (at least here in the states) |
| 20:52.24 | rrittgarn | (for true outbound calling) |
| 20:52.42 | devlap | Hmm. It looks like I got the double nat for this working. The only problem is, I'm not seeing numbers dialed from my phone when I WaitExten |
| 20:52.58 | Penguin | Yeah, the name is typically looked up by the recipient rather than received with the call. |
| 20:55.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:55.31 | Katty | i blame mister fender. |
| 20:57.23 | fabiobik | humm |
| 20:59.29 | MaliutaLap | Katty: good call |
| 21:00.33 | fabiobik | but for example, imagine that i buy something like this http://image.made-in-china.com/43f34j00KSFtmEponacq/Bulk-SMS-16-Port-GSM-Modem-Pool.jpg |
| 21:00.44 | fabiobik | and fill with gsm sim cards |
| 21:00.58 | fabiobik | is still possible to send text as callerid? |
| 21:04.52 | MaliutaLap | fabiobik: do you mean send non-numerical chars in CALLERID? or send SMS as callerid? |
| 21:05.10 | fabiobik | MaliutaLap, non--numerical chars |
| 21:05.21 | MaliutaLap | fabiobik: because the first is possible - although most ITSP's will disregaurd it on outgoing calls |
| 21:05.53 | MaliutaLap | fabiobik: you can re-write the CALLERID string based on DB lookups or the like |
| 21:06.28 | fabiobik | MaliutaLap, what i see in my voip provider for example, i can only get a number (if verify my phone number) |
| 21:07.23 | MaliutaLap | fabiobik: http://www.voip-info.org/wiki/view/Asterisk+variables |
| 21:07.39 | MaliutaLap | fabiobik: are we talking incoming calls? |
| 21:08.05 | fabiobik | MaliutaLap, we are talking about send sms |
| 21:09.13 | MaliutaLap | fabiobik: I'm pretty sure that you'd need to talk to your SIP provider about what they do with the callerid - they probably ignore what you put in there and set it to something else |
| 21:09.22 | MaliutaLap | fabiobik: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CALLERID for * 11 |
| 21:10.01 | *** join/#asterisk ThatDamnRanga (~wiretap@unaffiliated/wiretap) |
| 21:10.14 | fabiobik | MaliutaLap, but forget voip in this moment |
| 21:10.36 | fabiobik | imagine that i buy a gsm modem to send sms via gsm networks |
| 21:10.52 | fabiobik | im still able to send non-numeric callerid? |
| 21:10.53 | MaliutaLap | fabiobik: even taking SIP out of it the GSM provider will probably set the callerid |
| 21:11.26 | MaliutaLap | fabiobik: from what I know it _is_ possible, but the issue still comes down to the provider - SIP or GSM |
| 21:11.52 | fabiobik | i guess in portugual i will not have luck |
| 21:12.02 | fabiobik | *Portugal |
| 21:12.07 | MaliutaLap | fabiobik: about the only place I think it won't get rewritten is on POTS |
| 21:13.58 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 21:15.42 | fabiobik | MaliutaLap, can i talk with you in private? |
| 21:16.33 | MaliutaLap | fabiobik: sure, but I do have other stuff to work on :) |
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| 21:24.10 | devlap | hmm my handset is blinking red |
| 21:24.20 | devlap | And I called VM, there is none. |
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| 21:49.26 | *** join/#asterisk jmmills (~jmmills@204.16.200.135) |
| 21:49.47 | jmmills | Is there a way to get a feature to act on both peer and self channels? |
| 21:50.12 | jmmills | It appears if you add a feature with two different names, same dtmf seq, each acting on a different channel that it does neither |
| 21:50.32 | devlap | Ok so I called in, left a VM, and the WMI light hasn't started blinking yet. If I "reboot" the phone, it will most likely start to blink, how can I get it to blink now or whats its deal? |
| 21:50.51 | giesen_ | anyone got some experience with realtime? trying to add a single peer in realtime but it won't show up (and yes I have rtcachefriends enabled) |
| 21:53.23 | [TK]D-Fender | devlap: I don't recall seeing your configs or debug... |
| 21:54.23 | devlap | Well, if I leave a VM, that should trigger a MWI right? How is that sent to the phone? |
| 21:54.38 | [TK]D-Fender | when you tell it to. |
| 21:54.45 | [TK]D-Fender | So far we don't see you doing so |
| 22:02.08 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
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| 22:03.35 | devlap | http://pastebin.com/m7XJ0eew is my sip conf for my device and the voicemail conf |
| 22:04.24 | devlap | Again though, how is that MWI sent to the phone when a voicemail is left? |
| 22:10.15 | rrittgarn | vmexten=mailbox@context i believe |
| 22:10.57 | rrittgarn | and in sip.conf subscribemwi either not set or set to no |
| 22:11.39 | Penguin | I've never used vmexten or subscribemwi, and MWI works. |
| 22:11.39 | rrittgarn | i usually set mailbox and vmexten... i've never taken the time to figure out which is necessary |
| 22:12.08 | devlap | does mailbox require the context like EXT/default? |
| 22:12.14 | devlap | Mine I just have EXT1 |
| 22:12.30 | Penguin | Yes, but if you omit the @default, it should mean @default. |
| 22:14.16 | devlap | ok ill try adding subscribe |
| 22:14.44 | Penguin | What kind of phone is it? |
| 22:15.10 | devlap | grandstream 2160 |
| 22:16.35 | jmmills | great http://www.voip-info.org/wiki/view/Asterisk+cmd+PauseMonitor |
| 22:16.43 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 22:16.47 | jmmills | Just drops the channel |
| 22:22.27 | [TK]D-Fender | [18:14]devlapok ill try adding subscribe <-- nope |
| 22:22.34 | [TK]D-Fender | mailbox =box@vmcontext |
| 22:22.41 | [TK]D-Fender | that is what you set to get your mwi |
| 22:23.36 | devlap | ok, its EXT1 now, I'l set EXT1@default. |
| 22:23.57 | [TK]D-Fender | EXT1? |
| 22:24.03 | devlap | yea. |
| 22:24.04 | [TK]D-Fender | this has to be the box # from voicemail.conf |
| 22:24.09 | [TK]D-Fender | not letters allaowed |
| 22:24.42 | devlap | EXT1 => 200,a user,a@test.local |
| 22:24.42 | devlap | <PROTECTED> |
| 22:24.55 | [TK]D-Fender | yo should not have letters in the mailbox # |
| 22:25.07 | [TK]D-Fender | this is not normally allowed |
| 22:25.17 | devlap | So it doesn't work like dialplan extensions? |
| 22:25.19 | [TK]D-Fender | no |
| 22:25.41 | [TK]D-Fender | they are nothing alike |
| 22:25.51 | devlap | I know they arent alike, they carry similar syntax it seems. |
| 22:26.03 | [TK]D-Fender | nope |
| 22:27.02 | [TK]D-Fender | voicemail is more like a CSV of mailbox lines. dialplan has priorities, applications, and all sorts of stuff parsed in a call. |
| 22:28.33 | devlap | I didn't say voicemail and dialplans, I said voicemail, referring to the voicemailbox number, and the dialplan extensions, which are declared via simiar syntax dialpanexten => definition , vs voicemailboxid => definition |
| 22:29.06 | devlap | Since you are allowed dialplan extensions with letters, I use them in some cases, it makes sense that vm could've worked similary |
| 22:29.14 | devlap | I changed over my extra set 1 though |
| 22:29.19 | devlap | to use 200 instead of EXT1 |
| 22:29.22 | devlap | will see what happens. |
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| 23:32.43 | giesen_ | anyone know how to test an ODBC command if the result is null (returned no rows)? |
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