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| 03:03.20 | Ellenor | eeeeeee |
| 03:03.23 | Ellenor | e_e |
| 03:36.59 | ChannelZ | aaaaaaaaaaa |
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| 09:09.16 | MajesticFudgie | On support lines when you select certain options how are they coveyed to the support staff on the other end? |
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| 09:36.05 | MajesticFudgie | I've also randomly encountered an issue where my desk phone isn't placing calls properly. Not sure if its Asterisk or the phone. |
| 09:36.22 | MajesticFudgie | I start the call, it rings then hangs up when ringing outbound through sipgate |
| 09:36.27 | MajesticFudgie | and in some cases it doesnt even ring |
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| 09:55.56 | plasmoduck | Hey I need some help, I just setup my Thompson ST2030 VoIP phone and made a call, the quality was cutting out, so I am wanting to change the codec setup, what should I try? |
| 09:56.04 | plasmoduck | For the following; G.711U G.711A G.723_63 G.729AB |
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| 11:27.02 | *** join/#asterisk Haris (~haris@unaffiliated/haris) |
| 11:29.31 | Haris | I have the default group "all users" in freepbx. For members of that group, I enabled webrtc. how can I check if the users or extensions links to them inherited that setting in "enabled" form ? |
| 11:30.43 | Samot | Haris: Asking FreePBX questions in here is pointless. |
| 11:31.29 | Haris | this is an asterisk Q with some detail from freepbx point of view |
| 11:31.53 | Haris | there are asterisk cli tool commands that dump out settings |
| 11:31.56 | Samot | Except FreePBX does things differently. |
| 11:32.28 | Haris | and I believe asterisk cli settings dump will show the base/raw settings configured by freepbx for asterisk |
| 11:32.52 | Samot | The question you asked is a FreePBX question, not an Asterisk question. |
| 11:33.09 | Samot | You want to know how to tell what FreePBX has done and applied settings. |
| 11:33.20 | Haris | there's a command that goes like sip show <something> which dumps config of extensions/users. I was referrring to that |
| 11:34.17 | Samot | peer. |
| 11:36.03 | Haris | I can do sip client <-> asterisk <-> sip client video calls through desktop sip client like microsip. now need to make those calls via webrtc (web socket based). need to see a one page dump of extension/user settings to find out where I stood against that objective |
| 11:37.03 | Samot | sip show peer <peer> |
| 11:37.36 | WIMPy | Has someone flooded the channel again? |
| 11:37.54 | Haris | ? |
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| 11:38.01 | Haris | there's no flooding going on |
| 11:42.21 | WIMPy | Looks like you did some flooding some time. |
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| 11:46.18 | Haris | that may be a multi-line copy/paste of log output |
| 11:46.32 | Haris | may have been 5-8 lines |
| 11:46.52 | Haris | that was .. perhaps .. some time last week |
| 11:52.00 | Haris | guys, how can I configure web socket based video calls through asterisk 13 ? |
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| 11:53.08 | Samot | Haris: |
| 11:54.09 | Samot | How a straight Asterisk install handles configuration is different than how FreePBX does it. |
| 11:54.17 | *** part/#asterisk monsterco (~monsterco@TOROON474AW-LP140-01-1279749862.dsl.bell.ca) |
| 11:54.34 | Haris | yep. I saw some of that |
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| 11:54.43 | Haris | but still don't understand it |
| 11:54.59 | Samot | OK, so if you ask someone using straight Asterisk how to do a configuration... |
| 11:55.06 | Samot | First, they don't use GUIs. |
| 11:55.14 | Samot | Second, the file structure will be different. |
| 11:55.47 | Samot | Third, any custom dialplan or changes you make you have to make sure FreePBX doesn't overwrite them later. |
| 11:56.04 | Haris | hmm |
| 11:56.50 | Haris | I'v made no dialplan(s) yet. haven't even touched that part yet. I just created two extensions, which auto-created two users and then I configured some settings for these extension(s)/user(s) |
| 11:57.10 | Samot | Through a GUI! |
| 11:57.15 | Haris | so far, I'm proceeding through freepbx |
| 11:57.16 | Haris | yes |
| 11:57.25 | Samot | That's not covered here. |
| 11:57.36 | Samot | So again, stop asking FreePBX questions in Asterisk. |
| 11:58.05 | Haris | ok |
| 12:00.46 | s8y | I experience some problems with fop2 recording. Is there a way to record entire conversation instead of from the moment I press record? |
| 12:01.13 | Samot | Turn it on when the call starts. |
| 12:03.26 | s8y | Samot: I need it for my switch opearator, They process loads of calls a day and only need to record few of them in a month. They can decide they would like to record a call once good few min in conversation |
| 12:05.40 | s8y | Samot: thought mixmonitor would let me do that |
| 12:05.50 | Samot | With start codes yes. |
| 12:06.02 | Samot | With star codes yes or really any code you tell it to use. |
| 12:09.13 | s8y | Samot: so how would it work? I configured queues to be mixmonitor enabled can get it recorded after call is bridged to queue agent, what do I do next? |
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| 12:13.03 | Samot | It would be in your dialplan. |
| 12:13.16 | Samot | And your featuremap configs. |
| 12:17.00 | wyoung | hey |
| 12:17.10 | s8y | Samot: ok, thank's I will look into featuremap. Is there a way dtmf won't be heard by caller? |
| 12:17.30 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 12:17.37 | Samot | The caller should hear their own DTMF |
| 12:17.54 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:18.02 | Samot | The recordings probably won't hear them like you would think. |
| 12:18.10 | Samot | To stop from know the tones that are presssed. |
| 12:18.18 | Samot | To stop from knowing the tones that are pressed. |
| 12:19.30 | s8y | Samot: ok I now have general idea how it could work. Would you be able to point me to some guide/sample config please? |
| 12:21.38 | Samot | I'm just going to google anything that i send to you. |
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| 12:24.10 | s8y | Samot: I already done some googling but could not find exactly what I was after (perhaps asking wrong questions) |
| 12:24.18 | wyoung | Samot: It doesn't really matter if they do know you pressed a button, they can't reproduce it there end, nor do they know what you are doing |
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| 12:24.52 | Samot | I understand that. |
| 12:24.52 | wyoung | Samot: just blame the cat for walking on your desk phone *shrugs* |
| 12:25.35 | Samot | I was talking about DTMF per RFC2833 |
| 12:25.58 | Samot | When you listen to a recording, you don't hear the tones of the caller when they press the DTMF. |
| 12:26.30 | wyoung | ah |
| 12:26.32 | wyoung | no idea |
| 12:27.03 | Samot | The caller should always hear their own DTMF because it's playing back in their speaker. |
| 12:27.10 | Samot | The callee won't. |
| 12:27.15 | *** join/#asterisk Hello71 (~Hello71@wikipedia/Hello71) |
| 12:27.33 | s8y | wyoung: I was told caller should not hear any tones (that's the requirement) alternatively I can do postprocessing and play message to agent (press 1 if you want last call recorded and remove delete wav file from dialplan) but its the message they are going to hear 300-500x a day. |
| 12:27.50 | [TK]D-Fender | s8y, "core show applicaiton dial" <--- |
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| 12:37.32 | s8y | [Tk]D-Fender: thanks I already call my queues with krwrx options. It makes things ready but doesn't actually do e.g. recording |
| 12:38.02 | [TK]D-Fender | "makes things"? |
| 12:40.21 | *** join/#asterisk ntz (~jdoe@static-84-42-228-122.net.upcbroadband.cz) |
| 12:40.24 | ntz | hello |
| 12:40.39 | ntz | I'd like to request only a consultation |
| 12:41.20 | ntz | I've made working somewhere a fax solution using hylafax, t3modem and asterisk that provides SIP carrier for T38 |
| 12:42.00 | s8y | [Tk]D-Fender: allow called party to enable parking, enable recording and so on, make things ready. |
| 12:42.06 | ntz | now I'd like to ask - I have a dialplan written now in static manner using AEL ... Can I somehow be fetching a phone numbers from LDAP ? |
| 12:43.49 | *** join/#asterisk WIMPy (~wimpy@85.183.95.26) |
| 12:49.31 | ntz | and more more techical oriented question - what is the `no such number' signal ? |
| 12:50.07 | [TK]D-Fender | ntz, realtime dialplan. It's in the book & on the WIKI. Go read up. |
| 12:50.12 | ntz | eg when I send fax to some number not in my dialplan it just behaves like if there ain't a peer to that number waiting for it in queue to appear |
| 12:50.26 | ntz | [TK]D-Fender: ok, thanks |
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| 12:50.59 | [TK]D-Fender | ntz, show us this call. we have no idea what you're talking about |
| 12:51.05 | [TK]D-Fender | your wording is confusing |
| 12:52.14 | ntz | ok, minute |
| 12:54.04 | ntz | [TK]D-Fender: I have for PoC this dialplan http://susepaste.org/view/raw/9576717 .... now only asking, what I shall to add to context fax_out() to just tell to the sender side some message like `this number doesn't exist' |
| 12:55.15 | [TK]D-Fender | <[TK]D-Fender> ntz, show us this call. we have no idea what you're talking about |
| 12:56.15 | ntz | [TK]D-Fender: I'm asking, how shall I adjust the fax_out context that it says to his T38 peer ``no such a number exists'' if I try to send fax on that number |
| 12:56.28 | [TK]D-Fender | If you want a message then PLAYBACK something |
| 12:56.44 | [TK]D-Fender | Look in your soudns folder to see if there is one, otherwise record one yourself |
| 12:56.44 | ntz | [TK]D-Fender: it's fax, not voice |
| 12:57.03 | [TK]D-Fender | no such a number exists <- this isn't a FAX thing |
| 12:57.14 | [TK]D-Fender | you don't tell a fax that "no number exists". |
| 12:57.18 | [TK]D-Fender | there is no such thing |
| 12:57.24 | ntz | hmm |
| 12:58.09 | ntz | so can I somehow propagate back to the sending machine (virtual fax machine) something that it won't put the fax to the sendqueue and abort it straight away ? |
| 12:58.25 | ntz | okay, maybe it's not a task for asterisk |
| 12:59.44 | [TK]D-Fender | have * not accept the call. |
| 12:59.51 | [TK]D-Fender | How do YOU determine that a number is not available? |
| 13:00.17 | ntz | [TK]D-Fender: as you can see, I have few of them statically in dialplan ... |
| 13:00.36 | [TK]D-Fender | nom, you don't |
| 13:00.43 | ntz | but the task for the future is to adjust that dialplan that it uses LDAP as a repository for numbers |
| 13:00.56 | ntz | if ("${EXTEN}"="111207") { ..... |
| 13:00.58 | [TK]D-Fender | you use "if" to compare" |
| 13:01.01 | [TK]D-Fender | that is NOT a match |
| 13:01.12 | [TK]D-Fender | <PROTECTED> |
| 13:01.24 | [TK]D-Fender | stop using "if" and make ACTUAL extensions to match them |
| 13:01.40 | ntz | hmm, ok, thanks for points |
| 13:01.44 | [TK]D-Fender | 111207 => |
| 13:01.57 | [TK]D-Fender | it matches.. because _x. DOES match |
| 13:03.06 | ntz | [TK]D-Fender: ok, let's leave this for now, I promise that I'll re-audit your suggestions earlier, it is now working for me completely fine |
| 13:03.21 | *** join/#asterisk luckman212_ (~luckman21@unaffiliated/luckman212) |
| 13:03.27 | ntz | I can send faxes correctly accross 5 machines now |
| 13:03.41 | ntz | s/earlier/later/ |
| 13:03.46 | [TK]D-Fender | If you want t.38modem to be told that it is NOT one of those #'s.. then you need to stop putting a wildcard match in like tthat |
| 13:04.06 | ntz | hmm |
| 13:05.05 | ntz | [TK]D-Fender: ok, if you look on fax_out ... if I'll send fax to some number not in there, it does Busy(); Hangup(); |
| 13:05.21 | [TK]D-Fender | You're still ACCEPTING the call |
| 13:05.29 | ntz | yeah |
| 13:05.34 | [TK]D-Fender | So DON'T |
| 13:05.59 | ntz | so please tell me, how shall I reject the call - I am not expert on asterisk, playing with it second day, I'm glad that I made it working somehow |
| 13:06.38 | ntz | [TK]D-Fender: hah, bless me sir, it's working on me on solaris moreover, you can hardly imagine the tortures I had to pass through |
| 13:06.48 | ntz | s/on/for/1 |
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| 13:26.49 | *** join/#asterisk Diego0481 (57e043ad@gateway/web/freenode/ip.87.224.67.173) |
| 13:27.02 | Diego0481 | Hi |
| 13:27.20 | Diego0481 | I am looking for asterisk developers |
| 13:27.44 | Diego0481 | Anyone around, do you know a good companies that do this |
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| 13:31.15 | *** join/#asterisk Diego0481_ (57e043ad@gateway/web/freenode/ip.87.224.67.173) |
| 13:31.34 | Diego0481_ | Sorry I have lost the connection |
| 13:31.47 | *** join/#asterisk plasmoduck (~cjg@103.27.224.198) |
| 13:32.04 | Diego0481_ | <PROTECTED> |
| 13:33.00 | Diego0481_ | do you now anyone plasmoduck |
| 13:33.35 | plasmoduck | nope |
| 13:33.42 | s8y | Diego0481_: there is #asterisk-consul irc channel have a look there |
| 13:35.01 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-zkknfzvpikokgyxe) |
| 13:35.01 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:35.03 | [TK]D-Fender | What do you actually need? |
| 13:41.29 | *** join/#asterisk kolko (~kolko@46.48.58.17) |
| 13:44.51 | wyoung | plasmoduck? |
| 13:45.13 | plasmoduck | yes? |
| 13:45.41 | wyoung | nn |
| 13:45.58 | plasmoduck | ? |
| 13:49.04 | Diego0481_ | I am looking to customise a version of asterisknow that we actually us for home automation |
| 13:49.38 | Diego0481_ | we would need someone to make us a customised GUI and also do some configuration inside of it |
| 13:50.36 | [TK]D-Fender | Go check out the asterisk-boz mailing list |
| 13:50.41 | [TK]D-Fender | Go check out the asterisk-biz mailing list |
| 13:51.44 | Diego0481_ | our developers are not experienced on linux nor asterisk now and we would need to make this changes for the engineers to access easily network configs |
| 13:52.13 | Diego0481_ | As long as branding it with our own name and logo |
| 13:53.05 | Diego0481_ | where should i check that, sorry |
| 13:55.16 | [TK]D-Fender | <[TK]D-Fender> Go check out the asterisk-biz mailing list |
| 13:55.21 | [TK]D-Fender | www.asterisk.org <- |
| 13:58.08 | Diego0481_ | Thanks |
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| 15:19.01 | mcargile | does asterisk need to be restarted to load changes made to rtp.conf? |
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| 15:42.29 | litn | hello, I'm trying to get CEL to log HOLD and UNHOLD events. I have events=all in my cel.conf, and I see most events, but there are no hold/unhold ones |
| 15:42.40 | litn | is it an app and not an event? |
| 15:43.30 | litn | asterisk 13.9 btw |
| 15:46.04 | litn | output of cel show status: https://pastebin.mozilla.org/8878399 |
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| 16:11.07 | igcewieling | litn: anything in sip.conf look like it could apply? |
| 16:13.18 | igcewieling | maybe some options related to call events? |
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| 16:26.57 | breadcrumbs | My calls are immediately hanged up... Extension is simple as exten => 999,1,Answer(). If i insert playback there, it works well, but then hangs immediately anyway. |
| 16:27.00 | breadcrumbs | what can i do? |
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| 16:28.53 | shido6 | show the call in pastebin - a sip debug would help considerably. |
| 16:31.40 | shido6 | ?pb |
| 16:31.41 | [TK]D-Fender | ~pb |
| 16:31.41 | 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. |
| 16:31.41 | breadcrumbs | call from console? |
| 16:31.41 | [TK]D-Fender | ^^^ |
| 16:31.41 | [TK]D-Fender | call from your phone |
| 16:31.41 | [TK]D-Fender | and SHOW us tthe CLI outtput |
| 16:31.41 | [TK]D-Fender | pastebin.com <- |
| 16:31.41 | breadcrumbs | aight, sec |
| 16:32.00 | breadcrumbs | ikr |
| 16:36.40 | breadcrumbs | shido6: http://pastebin.com/CRCpf0bd just this. |
| 16:37.05 | breadcrumbs | maybe Answer() is not enough... |
| 16:37.30 | [TK]D-Fender | your call fell through.. you have no more steps to process <- |
| 16:37.46 | [TK]D-Fender | <PROTECTED> |
| 16:37.50 | shido6 | So what do you want to do AFTER the call is answered? |
| 16:37.57 | [TK]D-Fender | You answered.. and that is all. Nothing more. The call ends |
| 16:38.00 | [TK]D-Fender | As it should |
| 16:38.21 | [TK]D-Fender | It will not just sit around for no good reason |
| 16:38.56 | [TK]D-Fender | Therefore "Answer" alone will answer the line, and then the call will end if you have nothing more to do. |
| 16:39.06 | breadcrumbs | Oh. I thought it just lets called finish call if he wants. |
| 16:39.13 | breadcrumbs | okay, i got it, thanks |
| 16:40.20 | [TK]D-Fender | nothing more to do = don't let them hold your lines hostage and run up your billing (if applicable) for no good reason. |
| 16:41.06 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-85-237.hsd1.or.comcast.net) |
| 16:54.11 | igcewieling | "I'd tell you a UDP joke, but you might not get it." |
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