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| 08:25.01 | Spajjie | Hi all |
| 08:25.30 | Spajjie | Does anybody know why my res_* modules would unload? (Currently running a FreePBX box) |
| 08:26.19 | tzafrir | Spajjie, unload? What do you mean? |
| 08:26.28 | tzafrir | Or do you mean: fail to load? |
| 08:27.04 | tzafrir | Any specific module? Also: which specific version of Asterisk? |
| 08:27.53 | Spajjie | load => res_ari.so |
| 08:27.53 | Spajjie | load => res_ari_applications.so |
| 08:27.53 | Spajjie | load => res_ari_asterisk.so |
| 08:27.53 | Spajjie | load => res_ari_bridges.so |
| 08:27.53 | Spajjie | load => res_ari_channels.so |
| 08:27.54 | Spajjie | load => res_ari_device_states.so |
| 08:27.54 | Spajjie | load => res_ari_endpoints.so |
| 08:27.55 | Spajjie | load => res_ari_events.so |
| 08:27.55 | Spajjie | load => res_ari_mailboxes.so |
| 08:27.56 | Spajjie | load => res_ari_model.so |
| 08:27.56 | Spajjie | load => res_ari_playbacks.so |
| 08:27.57 | Spajjie | load => res_ari_recordings.so |
| 08:27.57 | Spajjie | load => res_ari_sounds.so |
| 08:28.29 | Spajjie | Version 13 |
| 08:30.33 | Spajjie | odule Description Use Count Status Support Level |
| 08:30.33 | Spajjie | res_ari.so Asterisk RESTful Interface 0 Not Running core |
| 08:30.33 | Spajjie | res_ari_applications.so RESTful API module - Stasis application 0 Not Running core |
| 08:30.33 | Spajjie | res_ari_asterisk.so RESTful API module - Asterisk resources 0 Not Running core |
| 08:30.33 | Spajjie | res_ari_bridges.so RESTful API module - Bridge resources 0 Not Running core |
| 08:30.34 | Spajjie | res_ari_channels.so RESTful API module - Channel resources 0 Not Running core |
| 08:30.34 | Spajjie | res_ari_device_states.so RESTful API module - Device state resour 0 Not Running core |
| 08:30.35 | Spajjie | res_ari_endpoints.so RESTful API module - Endpoint resources 0 Not Running core |
| 08:30.35 | Spajjie | res_ari_events.so RESTful API module - WebSocket resource 0 Not Running core |
| 08:30.36 | Spajjie | res_ari_model.so ARI Model validators 0 Running core |
| 08:30.36 | Spajjie | res_ari_playbacks.so RESTful API module - Playback control re 0 Not Running core |
| 08:30.37 | Spajjie | res_ari_recordings.so RESTful API module - Recording resources 0 Not Running core |
| 08:30.37 | Spajjie | res_ari_sounds.so RESTful API module - Sound resources 0 Not Running core |
| 08:30.38 | Spajjie | 12 modules loaded |
| 08:30.48 | Spajjie | (Oops sorry for the spam) |
| 08:38.44 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:41.53 | slima | I need to: dial to SIP/1 (but if SIP/1 have a call, dial to SIP/2) call-limit for SIP/1=4 so the line is not 'busy'. How to determine state of line, before dial? Or how to do that another way? |
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| 09:15.59 | pawiecki | Spajjie: ~paste |
| 09:16.09 | pawiecki | ~paste |
| 09:16.09 | infobot | extra, extra, read all about it, paste is http://pastebin.org/ or http://bin.cakephp.org/ or http://pastebin.ca/ |
| 09:16.31 | tzafrir | Spajjie, maybe you merely miss some modules? |
| 09:17.03 | tzafrir | Spajjie, Can you try loading all modules (auto=yes or whatever it is called)? |
| 09:17.56 | tzafrir | Spajjie, If this helps, it means you need to start isolating the missing module(s) - just start with a list of all modules, and start removing modules from it. |
| 09:18.46 | tzafrir | slima, if you have call-limit enabled, asterisk should do that for you, I believe |
| 09:19.06 | pawiecki | slima: I'm not sure what is the recommended way, but I would simply set up phones, so there's no callwaiting, then just dial - if you get Busy, you can dial another device. |
| 09:19.19 | tzafrir | So just use Dial(SIP/1&SIP/2) |
| 09:19.50 | tzafrir | errr... right. of course what I wrote is wrong. |
| 09:22.56 | slima | I have call limit = 4 for this account, but I want to write dialplan if SIP/1 is not in 'use' state dial for it, else dial SIP/2 |
| 09:23.27 | tzafrir | Anyway, use channel originate Local/600@demo application Dial SIP/1 to saturate SIP/1 with calls |
| 09:24.06 | tzafrir | See what pawiecki wrote. |
| 09:27.29 | slima | I dont want this approach. |
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| 10:27.39 | tzafrir | slima, what's wrong with this approach? |
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| 10:42.34 | slima | tzafrir: because I like to have 'call waiting' feature elsewhere for this account. |
| 10:43.05 | slima | I use Group: same => n,Set(GROUP()=31) same => n,GotoIf($[${GROUP_COUNT(31)} > 1]?s,dorota) same => n,Dial(SIP/31,20,tm) same => n(dorota),Dial(SIP/13,20,tm) |
| 10:51.12 | pawiecki | slima: please use paste |
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| 12:25.35 | Samot | Uhm, why didn't any one mention DEVICE_STATE or EXTENSION_STATE? |
| 12:25.45 | Samot | Either one of those could be used to check the current state |
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| 12:37.38 | pawiecki | https://wiki.asterisk.org/wiki/display/AST/Querying+and+Manipulating+State |
| 12:37.46 | Dirk23 | Hi there. Does anybody know a good tutorial for recieving Fax with FreePBX? |
| 12:41.42 | davidbowlby | Dirk23, https://wiki.freepbx.org/display/FPG/Fax+Configuration |
| 12:41.54 | davidbowlby | LMGTFY |
| 12:42.06 | Samot | well.. |
| 12:42.22 | Samot | Could have started by asking that question in #freepbx |
| 12:43.58 | davidbowlby | Dirk23, it also depends on what hardware you're using |
| 12:45.08 | Dirk23 | davidbowlby: i found that too, i didnt know if that really is all i need to do! |
| 12:45.28 | davidbowlby | Dirk23, are you using hardware? |
| 12:45.43 | Dirk23 | Hardware? |
| 12:46.17 | Samot | Jesus. |
| 12:46.23 | [TK]D-Fender | phyiscal lines on-site using a DAHDI-based card in your server |
| 12:46.24 | davidbowlby | laughs |
| 12:46.32 | Samot | Dirk23: Are you looking to do inbound faxes to email? |
| 12:46.40 | Dirk23 | Sure i have hardware. did you ever see a Computer without it? |
| 12:46.51 | Samot | Dirk23: Or are you looking to use a device connected to a fax machine? |
| 12:46.51 | Dirk23 | Samot: yes. wRE ONLY |
| 12:46.54 | [TK]D-Fender | Dirk23, physical lines on-site using a DAHDI-based card in your server |
| 12:46.55 | Dirk23 | SORRY |
| 12:46.57 | davidbowlby | Dirk23, this is a fax module, we'll do the jokes |
| 12:47.03 | Dirk23 | we only have SIP |
| 12:47.07 | Samot | OK |
| 12:47.10 | davidbowlby | WE?? |
| 12:47.12 | Samot | So you want Fax2Email |
| 12:47.18 | Samot | Hush, he's talking about his company |
| 12:47.19 | Dirk23 | Samot: yes |
| 12:47.20 | Samot | I get it. |
| 12:47.21 | Samot | OK |
| 12:47.29 | Samot | Then follow the instructions in the wiki |
| 12:47.36 | davidbowlby | Samot, I know, so there should be someone who does this, right? |
| 12:47.58 | Samot | The Distro has all the needed software already installed for what you want to do. |
| 12:48.26 | Dirk23 | so i need no extension for that fax number, just following that wiki? No Inbound Route? |
| 12:48.37 | Samot | Uh. |
| 12:48.43 | Samot | Read the wiki again |
| 12:48.46 | Dirk23 | ok |
| 12:48.57 | Samot | The Inbound Route is WHERE you tell it to look for faxes for fax2email. |
| 12:49.30 | davidbowlby | Go into a extension and enable fax and set the fax to email address. |
| 12:49.30 | davidbowlby | Then in your inbound route pick the destination of fax recipient and the user. |
| 12:49.44 | Samot | Well, User Manager. |
| 12:49.57 | Samot | Since "FAX User" are linked to users not devices. |
| 12:50.16 | davidbowlby | Samot, I'm purely going on google searches, i don't use GUIs |
| 12:50.38 | Samot | OK, then just state that you don't use FreePBX |
| 12:50.47 | davidbowlby | which is what Dirk23 could have done on his own... |
| 12:50.53 | Samot | Like I said. |
| 12:51.00 | Samot | He could have asked in #freepbx |
| 12:51.19 | Dirk23 | sorry for bothering |
| 12:51.31 | davidbowlby | I suppose if we all look back at our lives, there are a lot of things we *could* have done differently |
| 12:51.34 | davidbowlby | but here we are |
| 12:51.35 | Samot | Dirk23: Where you do you think you'll get more detailed answers? |
| 12:51.40 | davidbowlby | talking to Dirk23 |
| 12:51.45 | Samot | In the channel dedicated to FreePBX or some other channel? |
| 12:52.10 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 12:55.38 | davidbowlby | I will say though Dirk23, I appreciate the question. I don't use fax really, but now you've got me adding a line to my lab |
| 12:58.56 | Dirk23 | Well i would like to have no fax, because its stinking old shit, but my Company needs it :P |
| 13:00.33 | Samot | Yeah well FAX isn't going away any time soon. |
| 13:00.48 | Dirk23 | Its like a zombi |
| 13:03.39 | fauxalliance | shudders, just take a snap and email it to me. |
| 13:04.03 | Samot | FAX has an expectation level. |
| 13:04.14 | fauxalliance | fwiw. most ricoh fax machines in an enterprise SCAN TO EMAIL. |
| 13:04.29 | fauxalliance | just sayin'. |
| 13:04.51 | Samot | FAX has an expectation level. |
| 13:05.29 | fauxalliance | sadly, expect little, get less. |
| 13:05.40 | Samot | OK. |
| 13:05.54 | Samot | FAX is point to point. |
| 13:06.06 | Samot | To a degree. |
| 13:06.13 | Samot | Email is not. |
| 13:06.19 | fauxalliance | i'm _not_ going there this morning. |
| 13:06.41 | Samot | Then you should have not said something. |
| 13:06.52 | Samot | Don't get involved in the conversation if you're not willing to have it. |
| 13:07.37 | fauxalliance | Samot: i agree with you most of the time, you're flavor of 'effective communication" is 20% what you know and 80% how you feel about what you know... |
| 13:07.41 | fauxalliance | i feel different on fax. |
| 13:07.46 | fauxalliance | end of story. |
| 13:07.52 | Samot | OK |
| 13:08.26 | Samot | Strangely... |
| 13:08.42 | Samot | My flavor of "effective communications" seems to work just fine. |
| 13:08.47 | fauxalliance | it does |
| 13:08.55 | fauxalliance | reminds me of Stallman |
| 13:09.31 | fauxalliance | that was an honest compliment, not a backhanded remark |
| 13:11.04 | *** join/#asterisk billxx (49958984@gateway/web/freenode/ip.73.149.137.132) |
| 13:12.16 | pawiecki | Samot: what do you mean saying Fax has an expectation level? |
| 13:12.38 | Samot | 9:05:53 AM <Samot> FAX is point to point. |
| 13:12.38 | Samot | 9:06:04 AM <Samot> To a degree. |
| 13:13.02 | Samot | When I send a FAX from my machine, the other side answers. |
| 13:13.23 | Samot | I know the other side has answered, I know they are getting the call and it's between me and them. |
| 13:13.30 | Samot | If I send an email... |
| 13:14.07 | Samot | It's got to get past my client, my email server, the Internet, the other sides email filters/spam/etc, their SMTP server and then into their mail client |
| 13:14.21 | Samot | Plus the email could be a forward that goes to, who knows. |
| 13:14.51 | Samot | Certain industry require an expectation of privacy or service level when sending faxes. |
| 13:15.12 | Samot | Some require certain things when receiving.. |
| 13:15.14 | davidbowlby | Samot, all that shows is that the fax is laying on someones desk |
| 13:15.15 | Samot | Or storing. |
| 13:15.30 | davidbowlby | you don't even know if it is the right desk |
| 13:15.32 | davidbowlby | or building |
| 13:15.35 | davidbowlby | or country |
| 13:15.40 | Samot | No do you with email |
| 13:15.42 | Samot | nor |
| 13:15.48 | Samot | So that argument is moot |
| 13:15.53 | davidbowlby | I disagree |
| 13:16.00 | davidbowlby | fax is a single number |
| 13:16.05 | davidbowlby | e-mail is a domain |
| 13:16.20 | Samot | Right. |
| 13:16.30 | Samot | You know what happens when you start to call forward your FAX number? |
| 13:16.34 | davidbowlby | so from the aspect of hijacking, it's harder to do that with a domain |
| 13:16.41 | Samot | The carriers start to tell you, service is not guaranteed at that point. |
| 13:16.44 | davidbowlby | or just plain getting it wrong |
| 13:16.45 | Samot | Because they lose control. |
| 13:16.55 | Samot | You're kidding right? |
| 13:17.06 | davidbowlby | no |
| 13:17.10 | Samot | OK. |
| 13:18.47 | pawiecki | Samot: thanks for explanation. |
| 13:18.59 | Samot | Np. |
| 13:19.17 | Samot | There's a reason FAX is still a prime form of communication in various industries. |
| 13:19.21 | Samot | Medical being one of the biggest. |
| 13:19.36 | *** join/#asterisk |stefan| (~stefan@65-233.soderhamn.com) |
| 13:19.38 | Samot | So until those billion dollar industries change their tune.. |
| 13:19.48 | Samot | FAX isn't on the verge of going any where. |
| 13:19.50 | fauxalliance | No... fax machines are taboo in our hospital. See Also HIPPA |
| 13:20.01 | Samot | Who have rules. |
| 13:20.02 | fauxalliance | send it to the wrong number, youre chagged |
| 13:20.03 | Samot | OK. |
| 13:20.04 | pawiecki | I found a tip on using fax if you need to - just mail encrypted and send fax with pass |
| 13:20.23 | |stefan| | Hey =) |
| 13:23.34 | |stefan| | I'm trying out PJSIP for the first time and i'm having some trouble getting calls out from my softphones . I've tried the troubleshooting guide and followed all the examples but somehow i'm still getting unauthorized and 488 when trying to call. registration of the softphone works though. but not calling =) |
| 13:24.33 | |stefan| | Using Asterisk version 13.14.1 |
| 13:24.49 | [TK]D-Fender | Show the failed attempt with SIP debug enabled |
| 13:24.56 | [TK]D-Fender | ~pb |
| 13:24.56 | infobot | hmm... 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:24.57 | [TK]D-Fender | ^^^ |
| 13:25.02 | |stefan| | alright. hang on =) |
| 13:27.22 | |stefan| | https://pastebin.com/AAcSHhzm |
| 13:29.25 | [TK]D-Fender | a=rtpmap:110 speex/8000 |
| 13:29.25 | [TK]D-Fender | a=rtpmap:97 iLBC/8000 |
| 13:29.34 | [TK]D-Fender | SIP/2.0 488 Not Acceptable Here |
| 13:29.43 | [TK]D-Fender | betting your codecs aren't matchign the offer.... |
| 13:30.16 | |stefan| | I've been thinking about that also. using zoiper and i have ulaw and alaw in the list but it's not being presented in the invite for some reason. |
| 13:30.32 | |stefan| | since i have disallowed all but ulaw and alaw on the user endpoints. |
| 13:31.30 | |stefan| | but. odd that i'm having the same issue with both android client and pc client with the codecs being presented in the invite. it works just fine with chan_sip. |
| 13:31.34 | [TK]D-Fender | this is from your softphone |
| 13:31.41 | [TK]D-Fender | go change the codecs and verify your * side |
| 13:34.38 | |stefan| | just.. help me think here for a bit =) 2 separate configurations on the server. when i'm using chan_sip i can see the codecs in the invite. but not when i'm using pjsip. and i change nothing on the actual softphones. |
| 13:35.11 | file | they are in the INVITE, they just have no rtpmap which is fine, I'd suggest providing your endpoint configuration |
| 13:35.41 | |stefan| | alright i'll provide an example. |
| 13:35.55 | [TK]D-Fender | <[TK]D-Fender> a=rtpmap:110 speex/8000 |
| 13:35.55 | [TK]D-Fender | <[TK]D-Fender> a=rtpmap:97 iLBC/8000 |
| 13:35.58 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 13:36.17 | file | they're in the m= line |
| 13:36.25 | file | as they are statically defined they don't need an rtpmap |
| 13:36.38 | file | it's only dynamic payloads that need an rtpmap in order for things to know what they are |
| 13:38.36 | |stefan| | https://pastebin.com/ZJFy95C4 |
| 13:38.39 | *** join/#asterisk voipmonk (~voipmonk@138.68.2.1) |
| 13:39.19 | [TK]D-Fender | And the softphone side.... |
| 13:39.35 | file | and xxx is ext? |
| 13:39.54 | |stefan| | the softphone is simple. xxx001 and pw and my hostname:port . nothing fancy. |
| 13:40.25 | file | and does pjsip show endpoint on the endpoint show the expected configuration? |
| 13:40.38 | |stefan| | afaik yes |
| 13:40.43 | |stefan| | i can give you a paste from that also. |
| 13:40.55 | [TK]D-Fender | <|stefan|> the softphone is simple. xxx001 and pw and my hostname:port . nothing fancy. <- this isn't a port issue |
| 13:41.02 | [TK]D-Fender | 488 is typically CODEC. |
| 13:41.08 | [TK]D-Fender | show us the settings |
| 13:41.29 | [TK]D-Fender | IDK why you thought confirming the IP & port mattered.... |
| 13:42.07 | |stefan| | all codecs are enabled in zoiper. ulaw alaw gsm speex ilbc30 and ilbc20 |
| 13:42.25 | |stefan| | like i said. it works with chan_sip with the "same" configuration. only allowing ulaw and alaw. |
| 13:42.29 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:42.29 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:42.47 | |stefan| | just trying to wrap my head around to why it ain't working =) |
| 13:43.02 | file | I just used that SDP against the latest 13 in a test and it worked as expected, no 488 |
| 13:45.06 | |stefan| | could there be some module dependency i've forgotten ? |
| 13:45.59 | file | doubtful, but providing the pjsip show endpoint would confirm the configuration |
| 13:46.09 | |stefan| | ok. hold. |
| 13:48.20 | |stefan| | https://pastebin.com/3xi8kcNN |
| 13:48.36 | file | is it ext001? |
| 13:48.41 | file | (are you sanitizing that) |
| 13:49.17 | |stefan| | heh. ye. |
| 13:49.37 | |stefan| | but it's probably pointless since i supplied it in my first pastebin link =) |
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| 13:50.05 | file | it all seems fine |
| 13:50.14 | |stefan| | i thought so too. |
| 13:51.47 | |stefan| | i'll try a different softphone just to test if it's an incompability issue with codecs. |
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| 14:02.58 | |stefan| | i'm getting the same from a different softphone when only enabling the allowed codecs also. |
| 14:03.25 | |stefan| | so question is if it's a version issue, or a configuration issue. possibly if i've left out some module that's needed. |
| 14:04.26 | file | are you being selective in modules? if so you'd need res_pjsip_sdp_rtp if you don't have it loaded... |
| 14:06.44 | |stefan| | i just tried it with autoload modules and it's still the same. |
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| 14:28.49 | |stefan| | when using pjsip. should i then disable loading chan_sip and res_rtp_asterisk modules ? |
| 14:31.59 | file | res_rtp_asterisk is used by both, chan_sip you can if you aren't using it or it can be run on a different port |
| 14:44.23 | *** join/#asterisk StucKman (~mdione@195.200.189.206) |
| 14:45.52 | StucKman | what do these message actually mean? SIP/jn-0008979b is making progress passing it to SIP/v4u7vi-0008979a |
| 14:48.10 | [TK]D-Fender | Means * got a progress status and is forwarding it |
| 14:48.42 | [TK]D-Fender | That is also the generic debug statement where you should actually be looking at the SIP debug |
| 14:49.02 | StucKman | ack |
| 14:49.37 | StucKman | and Probation passed - setting RTP source address to 85.31.193.194:6748 ? |
| 14:49.48 | [TK]D-Fender | Everything is fine |
| 14:49.56 | StucKman | for me probation is when they release you early |
| 14:49.59 | [TK]D-Fender | why are you tripping up on these? |
| 14:50.00 | StucKman | :-P |
| 14:50.14 | StucKman | just looking at logs, trying to understand them |
| 14:50.22 | file | it means that Asterisk determined the source of media so it can block any packets from other sources |
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| 15:07.33 | StucKman | I see this log: but no tthe 180 ringing SIP message: [Oct 26 15:08:09] VERBOSE[9026][C-0004811b] app_dial.c: [Oct 26 15:08:09] -- SIP/jn-0008979b is ringing |
| 15:11.20 | [TK]D-Fender | we don't see anything.... |
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| 15:31.49 | StucKman | [TK]D-Fender: uhm? I see the log, but our SIP message db (homer) can't find it |
| 15:32.18 | StucKman | so I'm not sure if the message is in response of a 180 ringing that the db didn't catch or is logger for another reason |
| 15:34.57 | [TK]D-Fender | * CLI |
| 15:35.08 | [TK]D-Fender | No need for anythign external |
| 15:36.12 | StucKman | uhm? sorry, I don't understand... |
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| 15:39.03 | [TK]D-Fender | * CLI |
| 15:39.07 | [TK]D-Fender | Forget logs |
| 15:39.47 | StucKman | you're telling me to use the asterisk CLI, ok |
| 15:40.06 | [TK]D-Fender | <[TK]D-Fender> * CLI <--------------------------- |
| 15:40.29 | StucKman | [TK]D-Fender: it's kinda terse as communication |
| 15:40.56 | StucKman | in fact, I'm trying to figure out something that happened to a call, and i only have logs and the SIP db |
| 15:41.28 | [TK]D-Fender | So this isn't something you can replicate on the spot? |
| 15:42.05 | StucKman | nop-e |
| 15:42.21 | StucKman | in fact, the agent called 1m after and it didn't occur |
| 15:42.32 | StucKman | I just need proof for my boss |
| 15:42.40 | StucKman | called again* |
| 15:51.17 | StucKman | also: I like to understand what the tolls I'm using are doing and how they report |
| 15:51.41 | StucKman | logs are a way to see what a toll has done |
| 15:51.53 | StucKman | anyways, ECANTREPRO, ticket closed |
| 15:52.06 | StucKman | thanks for the info |
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| 17:36.01 | sibiria | is there a way to context-wise limit the simultaneous number of calls asterisk will keep running? |
| 17:36.37 | sibiria | perhaps by "tagging" call file and somehow limiting per tag or so |
| 17:38.28 | sibiria | we currently achieve this by controlling how much we feed the outgoing spool, but it's a bit of a clunky solution in a way |
| 17:46.51 | [TK]D-Fender | "core show functions like GROUP" <- |
| 17:52.22 | sibiria | exactly what i was looking for. thanks, [TK]D-Fender |
| 18:04.08 | slima | If I have: sip <-> asterisk <-> sip to phone adapter <> analog fax machine setup, Is there any special config on asterisk required? |
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| 18:08.03 | [TK]D-Fender | T.38 <- |
| 18:08.18 | [TK]D-Fender | Make sure your provider supports it and so does your ATA and configure your peers accordingly |
| 18:30.05 | sibiria | silly question, but is there a way in the dial plan to return the call to the spool so to speak, without hanging up which causes it to be finalized? |
| 18:30.12 | sibiria | outbound calls |
| 18:36.04 | [TK]D-Fender | There is nothng to "return" to. |
| 18:36.10 | [TK]D-Fender | I don't know where you're going with this |
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| 18:40.14 | sibiria | to simply abort the outbound call and let the call file remain in the outgoing spool for another try at a later point (perhaps as per the MaxRetries/RetryTime settings) |
| 18:41.48 | [TK]D-Fender | no. |
| 18:42.06 | [TK]D-Fender | You should only be in the dialplan because it answered |
| 18:42.13 | [TK]D-Fender | there is never any going "back" |
| 18:42.39 | [TK]D-Fender | unless you are doing something out of the norm |
| 18:44.32 | sibiria | unless i'm mistaken, isn't it possible to initiate a dial plan context another way and from there do the actual dialing? |
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| 18:45.04 | sibiria | i could misremember but i recall seeing some example starting a macro from the call file, which in turn called Dial() |
| 18:45.14 | [TK]D-Fender | yes if your Channel: is a Local channel |
| 18:45.16 | sibiria | rather than ending up in the dial plan once the call was answered |
| 18:45.30 | [TK]D-Fender | then it folows the same ruels as a more direct tech as far as repeats,e tc go |
| 18:50.25 | sibiria | the GROUP functions were neat but i suppose there's no suitable way to manage contextual outbound call limits within asterisk itself |
| 18:50.54 | sibiria | unless i'd fire up a thousand outbound calls at the same time and let them monitor some global parameter |
| 18:51.09 | sibiria | seems a bit brutal |
| 19:02.18 | [TK]D-Fender | You should be writing a queuing script to issue them for you |
| 19:02.24 | [TK]D-Fender | in order to rate-control them |
| 19:02.47 | sibiria | it's what we're using now. the software itself is not so clunky, it's the method itself i'd like to be able to control in asterisk itself |
| 19:03.10 | [TK]D-Fender | * doesn't give you that option |
| 19:03.15 | [TK]D-Fender | You have to manage it yourself |
| 19:03.33 | sibiria | ideally we'd like a way to have the outgoing spool directory digest at set rate, the way we do now by just moving files there at a set rate |
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| 19:32.28 | threebar | Hey everyone, I have a q about Asterisk. I'd like to set up a voice prompt system that answers a VOIP line and gives prompts to users. What VOIP service do I need to set this up in Asterisk? |
| 19:33.15 | [TK]D-Fender | You don't need any |
| 19:33.28 | [TK]D-Fender | if you want to use an ITSP for PSTN connectivity yuo certinaly can. |
| 19:33.42 | [TK]D-Fender | And can choose any that use a protocol that * supports. Most support SIP. |
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| 23:23.34 | |stefan| | I finally found out what was preventing me from making calls =) avpf was being an ass. disabled that and now it's working. |
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| 23:48.16 | file | ah yeah, that'd do it |
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