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| 10:11.52 | vadim1024 | hello i need some help with chan_dahdi.conf |
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| 10:48.52 | vadim1024 | hello i need some help with chan_dahdi.conf |
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| 11:10.45 | Samot | vadim1024: I don't use DAHDi but you're better off actually detailing the issue you are having. That way when someone checks in on the channel they can see the issue and might be able to help. |
| 11:13.19 | vadim1024 | i have 2 T1 cards and here is my chan_dahdi.conf: |
| 11:13.31 | vadim1024 | [channels] |
| 11:13.33 | vadim1024 | <PROTECTED> |
| 11:13.35 | vadim1024 | <PROTECTED> |
| 11:13.37 | vadim1024 | <PROTECTED> |
| 11:13.39 | vadim1024 | <PROTECTED> |
| 11:13.41 | vadim1024 | <PROTECTED> |
| 11:13.43 | vadim1024 | <PROTECTED> |
| 11:13.45 | vadim1024 | <PROTECTED> |
| 11:13.47 | vadim1024 | <PROTECTED> |
| 11:13.49 | vadim1024 | <PROTECTED> |
| 11:13.51 | vadim1024 | <PROTECTED> |
| 11:13.53 | vadim1024 | <PROTECTED> |
| 11:13.55 | vadim1024 | <PROTECTED> |
| 11:14.12 | vadim1024 | I want to merge all channels into one group |
| 11:14.46 | Samot | Don't flood the channel. |
| 11:14.49 | Samot | ~pb |
| 11:14.49 | infobot | somebody said pastebin was 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 11:15.33 | vadim1024 | Sorry, wilco next time |
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| 11:38.36 | vadim1024 | hello, i have 2 T1 cards, here is my chan_dahdi.conf: http://pastebin.com/rZDc42fH and i want to merge all channels in one group... How i should modify my conf file? |
| 12:20.31 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
| 12:21.07 | [TK]D-Fender | vadim1024, change the group # |
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| 12:29.06 | Samot | See, someone would come along and be able to help. |
| 12:38.47 | vadim1024 | [TK]D-Fender You mean to change group = 2 to group = 1. |
| 12:38.51 | vadim1024 | ? |
| 12:40.06 | [TK]D-Fender | You want them in the same group.... so it'd be nice if you picked the same group.... |
| 12:40.11 | WIMPy | Or just remove it. |
| 12:40.37 | vadim1024 | [TK]D-Fender, ok thanks |
| 12:40.47 | WIMPy | As settings are inherited until changed, you can remove everything between the two channel lines. |
| 12:41.02 | [TK]D-Fender | yup |
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| 13:05.38 | tirej | hi everyone |
| 13:10.58 | *** join/#asterisk tikun (~David@luna.ellipse.net) |
| 13:15.00 | tikun | I have three POTS lines on a server I just recently got working to route all inbound calls on those physical lines over to the existing pbx via an IAX2 trunk, that was simple.. now I need to utilize those lines for outbound calls as well, how exactly would I set up an external pbx to use those phone lines for outbound? |
| 13:16.30 | tikun | I have a trunk atm configured to utilize dahdi channel 1, and what not, just not sure how I'd connect an outbound call over the IAX2 trunk, to then connect to that outbound trunk. |
| 13:17.34 | ruied | I'm trying to make a cron job to play a message hourly to asterisk sip phones (with paging). I'm trying from cli to start with. I can do "channel originate sip/23 application Playback tt-monkeys" directly to a sip phone (so I can make a cron later on) but do not know how I can set an Alert Info at cli originate before, so the phone automatically answers the phone... |
| 13:17.40 | WIMPy | Every call has an inbound and an outbound leg. The peers may swaitch roles, but it's always the same. |
| 13:18.18 | WIMPy | ruied: Use the dialplan. |
| 13:25.18 | [TK]D-Fender | And dial that as a Local channel |
| 13:25.57 | ruied | WIMPy, I have tried that and it works partially. From cli asterisk is reporting error from /dev/desp (that I don't need). Tryed also to /dev/null. I only need one way sound from asterisk (playing tt-monkeys for example) to the paging phones. |
| 13:26.10 | tikun | ok, so just create an inbound route on the iax/pots server, and have it match the incoming DID and then just dial out using the dahdi trunk with the correctly channel? |
| 13:27.18 | tikun | with the correct channel** |
| 13:28.12 | [TK]D-Fender | ~freepbx |
| 13:28.12 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 13:28.13 | [TK]D-Fender | ^^^ |
| 13:31.17 | ruied | the problem seems that asterisk is expecting to receive sound and it keeps reporting error. Might need one way calling/audio or something like that... |
| 13:32.14 | WIMPy | Not sure what you're talking about. |
| 13:34.37 | [TK]D-Fender | "seems that asterisk is expecting to receive sound and it keeps reporting error" <- beyond this you haven't told us about an actual problem. |
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| 13:36.42 | ruied | [TK]D-Fender, that is the only problem. after Dial the paging extension, asterisk keeps reporting the error until I restart it. And after the first paging I can't do another one until I restart *. http://pastebin.com/uy4ebmU2 |
| 13:36.56 | ruied | Sorry, my english is not very good... :) |
| 13:37.21 | ruied | And * also not very good... :) |
| 13:38.29 | [TK]D-Fender | Your approach also makes no sense |
| 13:38.37 | [TK]D-Fender | Who told you to use console dial? |
| 13:38.41 | [TK]D-Fender | None of us did |
| 13:39.10 | [TK]D-Fender | Using Page() for a single device is pointless as well. |
| 13:39.26 | [TK]D-Fender | And you have 2 extens at priority 1 |
| 13:40.16 | [TK]D-Fender | There is no part of that pastebin that is right. |
| 13:40.23 | Rasputin3711 | Hangup() and Hangup(17) is the same? or what is the default code for Hangup() |
| 13:40.49 | Samot | What do you mean default code? |
| 13:41.04 | Rasputin3711 | http://wiki.freepbx.org/display/FOP/Hangup+Cause+Codes |
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| 13:41.49 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:42.21 | Samot | So you want to issue a hangup that is? |
| 13:42.26 | ruied | It's a first try. I would like to make a cron job at 18:00 asterisk sends a page to a group of phones with a Playback "Office closed". I have tried with console dial, so later I could set a cron job with something like /usr/sbin/asterisk -rx "console dial D_D@root" |
| 13:42.26 | Samot | There is no "default code" |
| 13:43.04 | [TK]D-Fender | ruied, ORIGINATE <- Or call files |
| 13:43.48 | [TK]D-Fender | There is no console. You aren't using a sound device on CLI as a softphone. "console dial" is NOT what you want |
| 13:45.46 | ruied | [TK]D-Fender, .call file makes sense.... |
| 13:45.52 | WIMPy | Rasputin3711: The default is 16 or whatever has been set by Dial(). |
| 13:46.07 | Samot | Well it depends on how the call is handled. |
| 13:46.16 | Samot | 16 is if the call is answered and hangs up. |
| 13:46.20 | Samot | Normal call clearing. |
| 13:46.26 | ruied | more sense than what I was trying... :) |
| 13:46.29 | Samot | But if the user is busy, it's going to be 18 |
| 13:47.12 | Rasputin3711 | GROUP_COUNT if more than 2, i want to hangup with busy |
| 13:47.21 | WIMPy | 17 |
| 13:47.23 | Samot | If you issue HangUp(18) it will hang up and send a 486. |
| 13:47.38 | Samot | Oh backwards. |
| 13:47.41 | Samot | 18 is 408. |
| 13:47.45 | Samot | No response. |
| 13:47.47 | [TK]D-Fender | just call Busy() then Hangup() |
| 13:47.51 | [TK]D-Fender | Should do it by itself |
| 13:47.55 | Samot | Can do that too. |
| 13:48.06 | [TK]D-Fender | #rounderwheels |
| 13:48.18 | WIMPy | No need to duplicate things. |
| 13:48.43 | Samot | But there is no default code because it's dependant on how the call was handled. |
| 14:05.20 | equilibrio | Anyone here familiar with CiscoPAGE and the cisco manager patch for asterisk? I am trying to only intercom one phone and be able to hear what the person says to me. Ciscopage goes only one way since its paging a lot of phones. Also, I cant setup an intercom line. FreePBX + cisco phones doesnt like that (a second line). As of now I have put that in the config files to page phones. Is there a command I can put to page and hear? ext |
| 14:05.48 | equilibrio | exten => 1011,1,SIPCiscoPage(1001&1002&1003&1004&1005,ov(75)d(From ${CALLERID(number)})) sorry |
| 14:08.30 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:08.31 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:11.42 | *** join/#asterisk DivideBy0x0 (~DivideBy0@unaffiliated/divideby0x0) |
| 14:12.11 | [TK]D-Fender | Where do you see this app at all? |
| 14:15.30 | equilibrio | CiscoPage? |
| 14:15.55 | [TK]D-Fender | yes |
| 14:16.06 | equilibrio | its patch from |
| 14:16.06 | equilibrio | http://docs.acsdata.co.nz/asterisk-cisco/document-overview.shtml |
| 14:16.08 | [TK]D-Fender | or the one you named in your sample |
| 14:16.08 | equilibrio | works good |
| 14:16.28 | equilibrio | the info is here: http://docs.acsdata.co.nz/asterisk-cisco/extensions-conf.shtml |
| 14:16.39 | equilibrio | show how to do unicast and multicast paging |
| 14:17.05 | equilibrio | it tells you you can add an intercom line to the cisco phone, but doesnt work, FreePBX doesnt allow the cisco phone to register 2 times (2 extensions) |
| 14:17.21 | Samot | Stop. |
| 14:17.25 | equilibrio | the intercom needs a phone to register its primary extension + an intercom line and both have credentials |
| 14:17.26 | Samot | That has nothing to do with FreePBX. |
| 14:17.36 | equilibrio | Samot: thats what I wanted :P to be stopped and know what im doing wrong |
| 14:17.38 | Samot | Chan_SIP doesn't like more than one registration. |
| 14:17.43 | equilibrio | Samot: ok :P |
| 14:17.57 | Samot | Understand how it works before determining the issue. |
| 14:18.05 | equilibrio | Samot: can I register a second line in any way |
| 14:18.23 | Samot | Yes, PJSIP. |
| 14:18.28 | equilibrio | Thing is we are a small shop and I do pretty much everything, everything is setup, everything works, just that intercom thing... |
| 14:18.29 | equilibrio | hmmm |
| 14:18.50 | equilibrio | ill try pjsip then, maybe it will solve the problem, but I remember having problem using it with those Cisco 9971 phones |
| 14:19.03 | [TK]D-Fender | Samot, No. |
| 14:19.04 | Samot | Oh. |
| 14:19.11 | equilibrio | but... |
| 14:19.13 | [TK]D-Fender | Samot, That's 2 DIFFERNT regs on the same phone |
| 14:19.17 | [TK]D-Fender | not the SAME account |
| 14:19.23 | Samot | Yeah, nevermind. |
| 14:19.31 | Samot | But really.. |
| 14:19.36 | equilibrio | lol |
| 14:19.36 | Samot | Get real IP phones. |
| 14:19.40 | [TK]D-Fender | <equilibrio> it tells you you can add an intercom line to the cisco phone, but doesnt work, FreePBX doesnt allow the cisco phone to register 2 times (2 extensions) <- Sur it does. |
| 14:19.58 | [TK]D-Fender | If your phone is registering properly then * doesn't care |
| 14:20.06 | equilibrio | hmmm |
| 14:20.09 | Samot | Yeah, I read that wrong. |
| 14:20.24 | equilibrio | should I put the credentials in the conf files? |
| 14:20.30 | Samot | Why? |
| 14:20.33 | Samot | You're using FreePBX. |
| 14:20.34 | equilibrio | but the intercom line and not make a new extension in freepbx |
| 14:20.36 | equilibrio | yes |
| 14:20.37 | Samot | That's what the GUI is for. |
| 14:20.46 | equilibrio | ok let me try it |
| 14:20.57 | Samot | Or get real SIP phones. |
| 14:20.59 | equilibrio | you gave me some faith :P |
| 14:21.01 | equilibrio | lol |
| 14:21.03 | Samot | That can do this without patches. |
| 14:21.13 | [TK]D-Fender | Telephony is not faith-based |
| 14:21.26 | Samot | Preach on. |
| 14:21.29 | equilibrio | its not me, its my boss who wanted cisco, we called cisco and they said for around 5000$ we should get a good system, we bought 5 phones and did the rest |
| 14:21.31 | equilibrio | cost 800$ |
| 14:21.33 | WIMPy | No, but VoIP is. |
| 14:21.41 | Samot | There are Cisco IP phones. |
| 14:21.47 | Samot | The SPA series. |
| 14:21.59 | equilibrio | yes I know, but he wanted that model lol |
| 14:22.01 | equilibrio | what can I do :P |
| 14:22.14 | Samot | Except you bought phones that are designed specifically for Cisco's UCM. |
| 14:22.22 | equilibrio | yup |
| 14:22.22 | Samot | That's why they wanted you to buy the whole system. |
| 14:22.28 | equilibrio | hence the cisco usecallmanager patch |
| 14:22.32 | equilibrio | yes I know |
| 14:22.40 | equilibrio | but still, 70$ per license and 60$ per voicemail |
| 14:22.42 | equilibrio | per phones |
| 14:22.44 | equilibrio | its a rip off |
| 14:22.46 | Samot | No one I have seen has yet to get these phones to operate at a 100% on non-UCM environments. |
| 14:22.51 | equilibrio | i understand how they make money now |
| 14:23.11 | Samot | Or you could have gotten a Cisco SPA5xx phone for like $150.. |
| 14:23.17 | Samot | And all this would already be supported. |
| 14:23.17 | equilibrio | Samot: I know, but I got all the buttons working good plus I can make apps in php, its all possible, but takes time |
| 14:23.27 | equilibrio | you are right |
| 14:23.57 | Samot | and you wouldn't be wasting time and *money* to save *money* on phones that don't work right with your setup. |
| 14:24.47 | equilibrio | Well video, paging, parking, voicemail, forward all, transfer, everything works, for what we need, just not the intercom which I am trying to fix :P |
| 14:24.56 | equilibrio | but yes, CUCM have much more features |
| 14:24.58 | equilibrio | thats for sure |
| 14:24.59 | Samot | "We saved $300 with these phones but it cost $400 to make them work. WIN!" |
| 14:25.04 | equilibrio | haha |
| 14:25.13 | equilibrio | well im not the one paying you know :P |
| 14:25.19 | Samot | transfer.. |
| 14:25.21 | Samot | voicemail |
| 14:25.24 | Samot | Those are no phone features. |
| 14:25.29 | equilibrio | video too |
| 14:25.44 | Samot | call forwarding is no a phone feature. |
| 14:25.52 | Samot | parking is not a phone feature. |
| 14:25.57 | equilibrio | anyways, i should try the spas yep |
| 14:28.31 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-nneqfjbzevsxobqf) |
| 14:31.08 | jrun | any oneliner to check if the rtp stream is passing through the server? |
| 14:31.35 | jrun | or it's run locally between endpoints on a particular lan. |
| 14:33.13 | Samot | Do you have directmedia set to yes or no? |
| 14:37.07 | *** join/#asterisk axisys (~axisys@unaffiliated/axisys) |
| 14:37.36 | jrun | yes |
| 14:38.02 | Samot | So then it will direct the media between the endpoints. |
| 14:38.07 | Samot | And no be involved. |
| 14:38.12 | Samot | And not be involved. |
| 14:38.44 | jrun | and see 'media will flow directly between them' |
| 14:38.59 | jrun | but i see the same thing when direct_media is set to no also! |
| 14:39.12 | Samot | What do you mean? |
| 14:39.35 | [TK]D-Fender | is no seeing configs or anything resembling proof |
| 14:39.36 | jrun | in console i see that msg saying, effectively, media is direct |
| 14:40.26 | *** part/#asterisk axisys (~axisys@unaffiliated/axisys) |
| 14:41.16 | jrun | regardless of direct_media being set to 'yes' or 'no' |
| 14:43.23 | [TK]D-Fender | is not seeing configs or anything resembling proof |
| 14:48.33 | *** join/#asterisk fullstop (~fullstop@107.170.53.155) |
| 14:48.48 | fullstop | Anyone else use voip.ms here? inbound calls are borked. |
| 14:49.55 | [TK]D-Fender | Show us |
| 14:53.14 | fullstop | they've acknowledged it now. https://voip.ms/m/feeds.php?feed=issuetracker |
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| 15:15.23 | DivideBy0x0 | fullstop: flowroute is saying the same thing |
| 15:15.43 | fullstop | wonder what happened.. |
| 15:19.15 | fullstop | vonage, flowroute, broadvoice, and voip.ms are all affected. |
| 15:21.52 | Nivex | I'm hearing scuttlebutt that it's an issue with Level3 |
| 15:22.20 | fullstop | https://www.reddit.com/r/networking/comments/55ttux/level_3_outage/ |
| 15:22.58 | Samot | It's L3 |
| 15:24.09 | fullstop | http://downdetector.com/status/level3/map/ |
| 15:27.05 | fullstop | the blobs keep growing.. haha |
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| 15:36.54 | Kobaz | aaaaaand it's down |
| 15:37.03 | Kobaz | if you zoom out most of the way |
| 15:37.08 | Kobaz | the blob looks like it's covering the entire US |
| 15:46.20 | Kobaz | i updated the map for you guys |
| 15:46.22 | Kobaz | http://i.imgur.com/ewohaSu.png |
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| 17:31.10 | scv | oh boy |
| 17:32.49 | scv | hitting some memory leak in one of 100 instances all running the same binaries, similar configs |
| 17:32.52 | scv | time to dig :/ |
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| 18:00.28 | drmessano | L3 probably powers their routers with Duracell batteries |
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| 18:22.54 | *** join/#asterisk mooasaurus (~gmann@c-71-200-232-20.hsd1.fl.comcast.net) |
| 18:24.14 | mooasaurus | hi, I'm looking for a way to forward a call to a number using a call file once the call is answered, is there a application available that would allow me to do this? |
| 18:26.34 | Samot | Why would you answer the call then forward it? |
| 18:27.54 | mooasaurus | is it necessary to go into why? I could think of a number of reasons |
| 18:28.08 | Samot | Well yes. |
| 18:28.24 | Samot | Because it would help understand what you are trying to do and if this is actually the best way to do it. |
| 18:28.45 | Samot | Because once a call is answered, it's answered. |
| 18:29.15 | scv | anybody have suggestions on tracking down this leak? |
| 18:30.31 | mooasaurus | call file is set up to dial out to a group number, once someone answers the call it will then forward it to a different group... eg. in the event you created a click-to-call script that called several people and wanted to be connected with whoever answered it |
| 18:31.00 | Samot | Why would it forward the answered call? |
| 18:31.33 | *** join/#asterisk matrix1233 (~matrix123@41.230.126.202) |
| 18:31.44 | Samot | Because you would initiate a Dial() that would call all the destinations.. |
| 18:31.49 | Samot | And the first one to answer it, gets it. |
| 18:32.34 | mooasaurus | wouldn't that defeat the purpose of a click-to-call script if you're wanting to forward the answered call to a different group number? |
| 18:32.35 | Samot | But why would you send me a call and when I answer it, take it away from me? |
| 18:32.48 | Samot | That's kinda mean. |
| 18:33.06 | Samot | Who answered the call?! |
| 18:33.30 | Samot | If you send a call to group1 with five people and one of them answers it, why would you forward it to group 2? |
| 18:33.33 | mooasaurus | I don't understand, if you create a call script and someone answers it, it will just hangup unless you do something else with it right? isn't that mean? |
| 18:33.42 | Samot | No. |
| 18:33.46 | Samot | That's not how it works. |
| 18:33.58 | Samot | Dial() sends the call. |
| 18:34.00 | Samot | I answer it. |
| 18:34.04 | Samot | The call is in progress. |
| 18:34.15 | mooasaurus | with who..? |
| 18:34.24 | Samot | OK. |
| 18:34.25 | mooasaurus | that is not how my asterisk box is set up |
| 18:34.34 | Samot | So you want to call two parties and bridge the call together. |
| 18:34.38 | mooasaurus | yes |
| 18:34.41 | Samot | Originate. |
| 18:34.50 | mooasaurus | I suppose, if we're on the same page |
| 18:34.57 | mooasaurus | originate is the application? |
| 18:35.01 | Samot | You use Originate to initial the call to the agent.. |
| 18:35.23 | Samot | Then you can play a "To connect" message.. |
| 18:35.30 | *** join/#asterisk bhans (~bhans@unaffiliated/bhans) |
| 18:35.36 | Samot | and dial the other party. |
| 18:35.59 | mooasaurus | so the call has to be initiated by the first party by pressing something? |
| 18:36.05 | Samot | No. |
| 18:36.08 | mooasaurus | it can't just automatically try the second number? |
| 18:36.13 | Samot | Yes. |
| 18:36.16 | Samot | It can. |
| 18:36.22 | mooasaurus | then why play the message? |
| 18:36.34 | Samot | I was just theorizing. |
| 18:36.49 | mooasaurus | ok so can this be done strictly from a call file? |
| 18:36.54 | Samot | Yes. |
| 18:37.06 | Samot | You tell it what context to execute when the call is answered. |
| 18:37.11 | Samot | So you dial the agent.. |
| 18:37.16 | Samot | They answer.. |
| 18:37.28 | mooasaurus | could you point me to a example? I would really appreciate it..I'm sorta new to asterisk |
| 18:37.29 | Samot | It executes the context, which can dial the other party. |
| 18:37.38 | Samot | wiki.asterisk.org |
| 18:38.33 | Samot | You click to call script will need to build the call file and then move it into the outgoing call spool. |
| 18:39.43 | mooasaurus | so for application, I would use originate, I can't use context again can I? |
| 18:40.13 | Samot | You can use AMI and the Originate command. |
| 18:40.26 | Samot | It will contain all the same details as the call file. |
| 18:40.53 | mooasaurus | AMI is not configured on my asterisk box, I thought you said this could all be done from the call file |
| 18:41.01 | Samot | It can be. |
| 18:41.28 | Samot | I just said, your click to call script will need to build the call file and then move it into the outgoing call spool. |
| 18:41.49 | mooasaurus | cool, I understand that part, just not the part of what actually needs to be in the call file to accomplish this |
| 18:42.04 | Samot | wiki.asterisk.org |
| 18:42.08 | Samot | search: callfiles |
| 18:42.33 | mooasaurus | this is why I'm asking for a example, because the wiki is not telling me, trust me I'm looking |
| 18:42.41 | Samot | Really? |
| 18:42.42 | mooasaurus | I'm reading the originate function right now |
| 18:42.45 | Samot | It's how I learned. |
| 18:42.49 | Samot | No, read callfiles |
| 18:42.55 | Samot | Look for callfiles |
| 18:43.55 | mooasaurus | sorry I was reading voip-info's page...let me look at this page |
| 18:44.35 | mooasaurus | so, I would use the originate application and data for it's arguments? |
| 18:45.28 | Samot | voip-info.org is outdated |
| 18:46.11 | mooasaurus | would the arguments be in a "blah,blah,blah" form factor in the call file? |
| 18:47.41 | mooasaurus | Data: sip/1234,app/ext??,etc |
| 18:49.55 | mooasaurus | still so confused..perhaps it's the terminology but I'm still not seeing how to accomplish this |
| 18:53.57 | [TK]D-Fender | forget "data". use Context,Exten,Priorirty |
| 18:54.11 | [TK]D-Fender | And forget Arguments while you're at it |
| 18:54.47 | [TK]D-Fender | it calls Channel: (just like AMI Originate). Nor args, etc. SO pick your channel type based on what you need to plan for (Local channels for headers, etc if required) |
| 18:54.59 | [TK]D-Fender | Tthen dumps the callee into the dialplan where you tell it to. |
| 18:57.19 | mooasaurus | can that accomplish a outbound call? |
| 18:57.32 | [TK]D-Fender | taht is what it does |
| 18:57.52 | mooasaurus | so extension would be the 11 digit number? |
| 18:58.25 | [TK]D-Fender | No, Extension: is where in the DIALPLAN the Channel: will be dumped once it answers |
| 18:59.18 | [TK]D-Fender | Ther is no assumption of what that value will be. |
| 18:59.54 | mooasaurus | yeah, so does it need to be configured in extensions.conf or something? |
| 19:00.03 | [TK]D-Fender | yes, that's what the dialplan is |
| 19:00.16 | mooasaurus | I'm trying to do this without any configuration what so ever other than the call file itself |
| 19:00.47 | [TK]D-Fender | Well * needs more files than just that obviously. |
| 19:00.49 | mooasaurus | because if I could get this working, the asterisk box would be serving more than just the purpose of this |
| 19:00.57 | [TK]D-Fender | you need to understand your dialplan basics |
| 19:01.13 | [TK]D-Fender | Not sto say you need anything commplex for this |
| 19:01.20 | [TK]D-Fender | Not to say you need anything complex for this |
| 19:01.26 | [TK]D-Fender | needs a new keyboard.... |
| 19:01.32 | WIMPy | Well, you don't NEED a dialplan for originate. |
| 19:01.34 | mooasaurus | I see that |
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| 19:02.29 | mooasaurus | yeah so how do I not use a dialplan with originate |
| 19:02.34 | [TK]D-Fender | calls files dial the Channel: (the same formatting as you'd put in a Dial()), then dumps them into the dilaplan. |
| 19:02.47 | [TK]D-Fender | If you want to take that first leg and call out ... then CALL OUT with another Dial() |
| 19:02.47 | WIMPy | By using an application instead of an extension. |
| 19:03.24 | [TK]D-Fender | You could maybe use Application:Dial direct, but you lose any followup capabilities completely. |
| 19:03.44 | mooasaurus | WIMPy: so you would specify multiple application: fields? |
| 19:03.49 | [TK]D-Fender | and ther is no point in losing that by starting that way when 1 simple pattern can get what you need if you';re using the same resourse to dial out, etc |
| 19:04.00 | [TK]D-Fender | <mooasaurus> WIMPy: so you would specify multiple application: fields? <- No. |
| 19:04.02 | WIMPy | No. There is no multiple anything. |
| 19:04.04 | [TK]D-Fender | Application is ONE |
| 19:04.13 | [TK]D-Fender | If you want more, then you use what I told you already |
| 19:04.52 | [TK]D-Fender | You have 2 choices : Single dialplan app, or get dummped to a specific extension in the dialplan |
| 19:05.30 | mooasaurus | and both require a dialplan pre-configured? |
| 19:05.42 | WIMPy | no |
| 19:06.01 | WIMPy | That's what I said. |
| 19:06.26 | mooasaurus | just figured I'd ask again because, I know you mentioned it, I'm still failing to see how it's done without a configured dialplan |
| 19:07.17 | mooasaurus | sorry for being ignorant, I've barely used asterisk, I just need it for this simple thing |
| 19:07.18 | WIMPy | We just told you. You send the channel to apllication Dial (or whatever). |
| 19:07.52 | WIMPy | "simple" and "Asterisk" to go along too well. |
| 19:07.53 | mooasaurus | so...application: Dial, data: number? |
| 19:07.57 | [TK]D-Fender | <WIMPy> By using an application instead of an extension. |
| 19:07.57 | [TK]D-Fender | <[TK]D-Fender> You could maybe use Application:Dial direct, but you lose any followup capabilities completely. |
| 19:07.59 | [TK]D-Fender | ^^^^^^^^^^^ |
| 19:08.07 | [TK]D-Fender | We've said it multiple times. |
| 19:08.19 | mooasaurus | YES but what you're failing to tell me is the specifics |
| 19:08.39 | [TK]D-Fender | 2 choices : Applicatio + Data, **** OR **** Context, Exten, Priority |
| 19:08.40 | WIMPy | data is the same you'd give to Dial() in the dialplan. |
| 19:08.41 | mooasaurus | you're saying application: dail, ok, where do I put the outbound number? |
| 19:08.57 | [TK]D-Fender | IN THE DATA |
| 19:09.07 | mooasaurus | COOL, thanks guys I will give that a shot |
| 19:09.12 | mooasaurus | appreciate the help |
| 19:12.34 | mooasaurus | worked perfectly, thank you |
| 19:33.36 | Kobaz | how do you set a sip error for hangup |
| 19:33.40 | Kobaz | you can do like, Hangup(cause) |
| 19:33.48 | Kobaz | but what about the actual 5xx or 4xx message |
| 19:34.04 | WIMPy | not |
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| 20:02.33 | testo | anyone know of any linux GUI dialer tools that can dial a number and then forward that call to an extension? |
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| 20:18.41 | [TK]D-Fender | https://www.google.ca/#q=asterisk+AMI+linux+GUI+dialer |
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| 20:28.50 | Kobaz | lmgtfy |
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| 22:44.29 | monsterco | Hi everyone; how can I upgrade from Asterisk 13 to 14? just do ./configure, make menuselect, make, make installm make config ? |
| 22:47.26 | gtjoseph | monsterco: Basically, that's it BUT you should read the UPGRADE.txt and CHANGES files to make sure something that you rely on hasn't changed. |
| 22:47.40 | gtjoseph | and of course...TEST |
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