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| 07:37.20 | Dirk23 | Hey. I have a FreePBX up and running. I am searching for a TAPI Driver for Windows or something like that, to connect my FreePBX with a CTI Software. Any suggestions? |
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| 09:32.28 | tafa2 | could anyone tell me how I could re-write caller ID's? |
| 09:33.01 | tafa2 | IF incoming caller ID = starts with 07, re-write to strip the leading 0 and prepend with 44 |
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| 09:40.16 | pawiecki | tafa2: Use Set and ${CALLERID(num)} variable? |
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| 09:41.43 | pawiecki | exten => 07X.,1,Set(CALLERID(num)=44${CALLERID(num):1}) - correct me if I'm wrong |
| 09:42.07 | vlt | Hello. In my dialplan I make an HTPP request for every incoming call. In the HTTP server logs I can see exactly when that happened. Now, hardly a single datetime matches with any in /var/log/asterisk/cdr-csv/Master.csv. Both servers' time is in sync. Any idea what might be a reason for this? |
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| 09:45.50 | vlt | vlt: UTC |
| 09:46.18 | vlt | vlt: Thanks ;-) |
| 09:46.47 | pawiecki | vlt: are you thanking yourself? :) |
| 09:48.31 | vlt | pawiecki: Yes :-) I just saved me a lot of time. |
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| 10:04.51 | tafa2 | pawiecki thanks bud |
| 10:05.07 | tafa2 | is that conditional on the fact that if the number starts with 07? |
| 10:05.23 | tafa2 | cos I've been looking at ExecIf |
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| 10:26.10 | pawiecki | tafa2: exten => 07X. will match any dialed number, that starts with 07 and is at least 3-digit. Check out the docs (they are really cool):https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 10:27.00 | pawiecki | https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 10:29.09 | pawiecki | also, ExecIf is fine, but I don't think you need it here. Try to keep things simple :) |
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| 10:29.56 | tafa2 | pawiecki |
| 10:30.01 | tafa2 | Dialled numbers is all good |
| 10:30.10 | tafa2 | I need to re-write *incoming* numbers |
| 10:30.30 | tafa2 | so if you call me from 07xxx I need to re-write it to 447xxx |
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| 10:46.25 | Marquel | morning. |
| 10:47.33 | Marquel | short question: is it possible to remotely hangup a calling channel? that is: a channel is calling an extension (thus Dial() is currently active) and now any extension executes a command that will send a Hangup() to that calling channel? |
| 11:03.21 | pawiecki | tafa2: sorry, i misunderstood that. You can use ExecIf or GotoIf with another context. For example: exten => s,1,GotoIf($[ "${CALLERID(num):0:2}" = "07" ]?set-new-cid:continue) |
| 11:04.15 | pawiecki | Marquel: yes |
| 11:07.59 | Marquel | pawiecki: you don't have a howto handy? |
| 11:08.29 | Marquel | pawiecki: tried PickupChan() with the channel of the incoming call, but that was "not found". |
| 11:08.37 | pawiecki | Marquel: Start with this => http://www.golinuxhub.com/2013/04/hanging-up-active-calls-in-asterisk.html and make your way into a working mechanism |
| 11:16.47 | pawiecki | Marquel: also, why do you want to hangup channels? |
| 11:18.39 | Marquel | pawiecki: suppose i have an external call ringing multiple internal extensions. now i do not want to take that call and get all internal extensions to get silent (noone else around to take the call) - so i want to "reject" that external call not only for "my" extensions but also on behalf of all others too. |
| 11:27.33 | pawiecki | Marquel: That doesn't feel right. Phones are supposed to ring, and people are supposed to answer them. Maybe set the ringtones to be more quiet or redo the dialplan so it's more suit to those types of situations (no people available to answer). |
| 11:29.21 | Marquel | pawiecki: trust me, when you are busy and your grandpa calls for the sixth time in five minutes with same issue and you really seriously have no time to answer, you want to reject that call. and in such a manner his phone tells him he's rejected. |
| 11:29.25 | pawiecki | You can for example change ringing strategy (ring one phone at a time, in a row) and duration (5 seconds and hangup). |
| 11:32.26 | pawiecki | Marquel: oh, I see, so that's more of a social than technical problem. Tell him to stop calling you or answer and Playback announce that tells him you're busy and to write sms/email for example. |
| 11:33.09 | Marquel | pawiecki: _if_ he would listen to "stop calling" i wouldn't have that problem. |
| 11:33.18 | Marquel | so i decided to take the technical approach. |
| 11:34.01 | pawiecki | Marquel: catch his number in the dialplan, use Answer and Playback with some announce, then Hangup. Problem solved. |
| 11:34.08 | Marquel | also, playback of such a recording (also good idea) would be technically the same thing as redirecting him to hangup(), wouldn't it? |
| 11:34.34 | Marquel | yeah, but that's too static ;) |
| 11:34.48 | Marquel | (doing that already for some other ... "customers"...) |
| 11:35.13 | pawiecki | no, Playback just plays audio file. There needs to be a next step, which could be Hangup. |
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| 11:38.35 | Marquel | pawiecki: yes, but getting the incoming call from the regular Dial() to the other track is still the same thing. |
| 11:39.18 | Marquel | so. ChannelRedirect() seems to be the way to go, just need to figure how to get the correct channel name. |
| 11:39.41 | pawiecki | Marquel: I don't know what a track is. |
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| 11:40.42 | Marquel | the line of applications... 1,Dial(...); 100,Playback(...);101,HangUp() and so on, 100 being the different track. |
| 11:57.49 | pawiecki | Marquel: that's not the "track", that's the priority of the same extension(s). I'm not sure what do you mean with ChannelRedirect() |
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| 11:59.44 | Marquel | pawiecki: ChannelRedirect picks up the incoming channel and puts it in context,exten,priority as desired. |
| 12:01.04 | Marquel | _if_ you know the name of the channel you mean ;) |
| 12:02.30 | pawiecki | Marquel: yes, but why not just use the GotoIF to check the CALLERID(num) ? |
| 12:02.39 | pawiecki | It's way simpler |
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| 12:18.19 | Marquel | pawiecki: because that's too static for my use-case. |
| 12:18.37 | Marquel | i just want to press a button just in case - sometimes i want to take the call ;) |
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| 12:26.30 | pawiecki | Marquel: then you can activate/deactivate this static rule with a code for example :) |
| 12:26.45 | pawiecki | **22 - active / **23 deactivated |
| 12:27.56 | Marquel | and forget it all the time... ;) |
| 12:28.16 | Marquel | and being unable to use it on other numbers. |
| 12:32.18 | [TK]D-Fender | Marquel, you need to ID the channel somehow and can then use CLI or AMI to hang up on them |
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| 12:33.29 | [TK]D-Fender | So dial something that checks for other channels from your device that are in state "ringing" and hangup the bridgechan |
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| 12:36.00 | [TK]D-Fender | Or if you aren't worried about multiple simultaneous calls you could jsut set a glabl variable or AstDB value before they dial you so you can have their channel handy immediately |
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| 12:36.58 | Marquel | [TK]D-Fender: going for the last option (fot the moment), need to figure the relevant name. currently thinking about CHANNEL(name), but haven't been able to test, yet. |
| 13:01.03 | znf | !ask |
| 13:01.18 | znf | oh wait, wrong channel |
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| 13:27.50 | pawiecki | What is a recommended way of sending Fax over IP, using T.38? I have a problem with Orange operator in France, who does not send me re-INVITE to change media from G.711 to T.38. They say, that I should send request to use T.38, when I send a fax call and they do not support any other option - are they right? |
| 13:32.56 | pawiecki | I've read this => https://tools.ietf.org/html/draft-mule-sip-t38callflows-00 but I'm not sure my interpretation is correct. |
| 13:37.05 | [TK]D-Fender | Saying what you read doesn't show what you did or how the other side reacted |
| 13:37.17 | [TK]D-Fender | Correct your approach |
| 13:37.48 | [NC] | pawiecki: It SHOULD be the recipient side that is responsible for re-inviting to T.38. So for an outbound call, it should be the telco that re-invites and for an inbound call, it should be your device. You can send re-invites if the telco doesn't and see if that helps, but it's also possible they just don't support T.38 to that destination, many telcos have T.38 available only to a subset of destinations... |
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| 13:44.10 | polysics | hey guys! Astricon was awesome and Asterisk 15 SFU is too |
| 13:45.00 | polysics | question for you: at Astricon 2016 there was a discussion about supporting streaming ASR and TTS services like Google or Watson. Did that eventually go anywhere? |
| 13:47.57 | pawiecki | [TK]D-Fender: ok |
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| 14:05.40 | pawiecki | [TK]D-Fender: example: https://pasteboard.co/GOKpIre.png |
| 14:07.19 | pawiecki | There's no re-INVITE and fax fails, because it's g.711. |
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| 14:12.03 | [TK]D-Fender | pawiecki, Show actual * SIP debug and actual configs |
| 14:17.46 | pawiecki | [TK]D-Fender: right now I don't have any debug to show you. The screenshot id from an older issue, that still is being discussed. I primarily asked to confirm what is the recommended way. This issue goes back and forth, and they are asking us again the same, already answered, questions, so I wasn't confident about it. |
| 14:17.51 | pawiecki | is* |
| 14:18.10 | pawiecki | [NC]: thanks. |
| 14:18.22 | [TK]D-Fender | Someone else's debug and lack of configs makes this a complete non-starter for me... |
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| 14:43.52 | Samot | pawiecki: What is the issue exactly? |
| 14:45.42 | pawiecki | Samot: When client was with another operator (polish Netia), we had no problems with calls and fax via T.38. Now, after client has changed operator, fax transmission fails when calling french numbers. |
| 14:46.42 | Samot | And now? |
| 14:46.46 | Samot | Who are they with? |
| 14:47.00 | pawiecki | Orange |
| 14:47.11 | Samot | So they changed ISPs? |
| 14:47.52 | Samot | You provide them with phone service? |
| 14:48.41 | pawiecki | Yes, Orange has their servers visible only from their network. |
| 14:49.11 | Samot | Huh? |
| 14:49.33 | Samot | What does the ISP have to do wit you giving them voip service? |
| 14:50.04 | pawiecki | My grammar sucks, I mean Orange's PBX is not routable from Internet, so they provide their own connection for that. |
| 14:50.20 | Samot | How are you giving them service? |
| 14:50.28 | Samot | So they changed ISPs and got a new PBX? |
| 14:51.01 | Samot | I'm trying to figure out how changing Internet providers broke your ability to give them T.38 |
| 14:51.04 | pawiecki | We only support their * PABX. |
| 14:51.08 | Samot | OK |
| 14:51.14 | Samot | That is not what I asked |
| 14:51.24 | Samot | 10:47:52 AM S<Samot> You provide them with phone service? |
| 14:51.35 | Samot | 10:48:41 AM P<pawiecki> Yes, Orange has their servers visible only from their network. |
| 14:51.42 | Samot | ^^ That's not " We just support the PBX" |
| 14:51.50 | Samot | The answer would have been "No, we just support the PBX" |
| 14:51.52 | pawiecki | define "phone service" then |
| 14:52.00 | Samot | The actual SERVICE |
| 14:52.07 | Samot | "Here's your sip trunk creds" |
| 14:52.13 | Samot | "You dial to this host" |
| 14:52.30 | pawiecki | That's SIP Trunk service for ya |
| 14:52.42 | pawiecki | so now define SIP Trunk service |
| 14:52.43 | Samot | But you don't provide the SIP Trunk. |
| 14:53.00 | Samot | Dude, do you give them DIDs? |
| 14:53.17 | Samot | Do they use your network to receive or make calls? |
| 14:53.33 | Samot | Or did they get their SIP trunk details from Orange? |
| 14:53.48 | pawiecki | phone service =! SIP Trunk service, so in my understanding supporting someone's PABX and their phones IS a phone service, so I answered "yes". You may argue with that, but question was far from being clear and specific. |
| 14:54.07 | Samot | Much like your description of this issue. |
| 14:54.21 | Samot | So you are the PBX ADMIN? |
| 14:54.23 | pawiecki | Samot: which description? |
| 14:54.46 | Samot | So Orange is the one this FAX has to go through |
| 14:54.49 | pawiecki | Samot: yes. |
| 14:55.15 | Samot | So now when the client faxes to France numbers, they fail? |
| 14:55.16 | pawiecki | Well, yes. That's what I meant, when I said, that they changed the operator to Orange. |
| 14:55.19 | pawiecki | What was not clear about it? |
| 14:55.28 | Samot | "Operator" is a person. |
| 14:55.33 | Samot | Or position. |
| 14:55.39 | pawiecki | Ok, then, "carrier is better? |
| 14:56.00 | pawiecki | yes, faxes fail |
| 14:56.05 | Samot | Well, it actually means something. |
| 14:56.06 | Samot | OK |
| 14:56.14 | Samot | So to all of France or just specific numbers? |
| 14:57.18 | pawiecki | All. |
| 14:57.36 | Samot | I find that hard to believe. |
| 14:57.46 | Samot | And no other FAX destinations have an issue? |
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| 14:58.54 | Marquel | is there a way to ask the calling channel its entire name (like SIP/bob-00001ab)? |
| 14:59.37 | Samot | ${CHANNEL} |
| 14:59.44 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables |
| 15:00.35 | Samot | pawiecki: Did the old carrier support T.38 and does Orange support T.38? |
| 15:03.52 | pawiecki | Samot: now i see a pattern here. I guess the info from client is incorrect. It's probably, that they send faxes to their |
| 15:04.31 | pawiecki | mother-company in France, and they also changed operator to Orange, so now the behaviour is different, because endpoints also changed operator. |
| 15:04.40 | pawiecki | But now I'm guessing. |
| 15:04.50 | Samot | pawiecki: Did the old carrier support T.38 and does Orange support T.38? |
| 15:05.16 | pawiecki | Yes, and yes. |
| 15:05.28 | Samot | And you're using T.38 on the trunk? |
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| 15:11.00 | pawiecki | For a fax call, first INVITE goes with G.711. If we receive reINVITE with T.38, then T.38 is used. Does that answer your question? |
| 15:11.39 | Samot | So you have T.38 set on the trunk? |
| 15:12.18 | pawiecki | Samot: how should I check, to be sure? |
| 15:12.29 | Samot | Look at your trunk settings. |
| 15:12.34 | Samot | Are you using Chan_SIP? |
| 15:13.51 | Marquel | okay, now i'm down to why the global variable appears empty inside an extension, after the dialplan has set it. |
| 15:17.25 | pawiecki | Samot: yes. |
| 15:17.48 | Samot | t38pt_udptl <-- Do you have that with any of it's options? |
| 15:17.56 | pawiecki | Marquel: did you reload the dialplan? |
| 15:18.05 | Samot | Either in the [general] section of sip.conf or in the peer section? |
| 15:19.54 | Marquel | pawiecki: yes. |
| 15:20.03 | pawiecki | Samot: I'm waiting for VPN to reconnect and will check, just a sec. |
| 15:20.24 | pawiecki | Marquel: show me how you defined variable |
| 15:20.25 | Marquel | pawiecki: the console logs the command yet dialplan show globals still show the variable as empty. |
| 15:21.00 | Samot | Marquel: Show the dialplan |
| 15:21.22 | Samot | Marquel: Show the logs from the Asterisk console for the call attempt. |
| 15:21.28 | Marquel | pawiecki: define or set? define is "[globals] VARNAME=", then there's exten => s,n,Set(VARNAME=${CHANNEL},g) |
| 15:22.12 | Samot | Marquel: Show the logs from the Asterisk console for the call attempt. |
| 15:22.17 | Samot | ~pb |
| 15:22.17 | infobot | extra, extra, read all about it, 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. |
| 15:22.45 | Marquel | <PROTECTED> |
| 15:22.47 | pawiecki | t38pt_udptl = yes, redundancy |
| 15:22.51 | pawiecki | Samot: ^ |
| 15:23.19 | pawiecki | I love the infobot |
| 15:23.22 | Marquel | (yes, i know of pastebin, not for single lines, though ;) ) |
| 15:23.27 | pawiecki | he's so cool |
| 15:23.44 | Samot | pawiecki: OK. So every other destination gets FAXes except for their central office in France? |
| 15:23.55 | Samot | Marquel: Show the logs from the Asterisk console for the call attempt. |
| 15:24.03 | Samot | ^^ As in more than one line. |
| 15:24.11 | Samot | You're in priority 8 of that context. |
| 15:24.21 | Samot | So there should be at least a few more lines. |
| 15:24.28 | Samot | Show the *whole* call attempt. |
| 15:25.37 | Samot | 11:21:28 AM M<Marquel> pawiecki: define or set? define is "[globals] VARNAME=", then there's exten => s,n,Set(VARNAME=${CHANNEL},g) |
| 15:25.46 | pawiecki | Samot: I'm not 100% sure, because we didn't test it so precisely. Also, can we continue this topic in about two hours or on monday? I should head home now, still at work. |
| 15:26.02 | Samot | If I'm around. |
| 15:26.03 | Samot | 11:22:46 AM M<Marquel> -- Executing [s@std-inbound:8] Set("SIP/inbound-00000169", "EXTERNALCHANNEL=SIP/inbound-00000169,g") in new stack |
| 15:26.13 | Samot | Marquel: Tell me how those match? |
| 15:26.16 | Samot | Because they don't. |
| 15:27.07 | pawiecki | Samot: ok, thanks for your input, bye all. |
| 15:28.08 | Marquel | https://pastebin.com/EzwUQXEs |
| 15:29.20 | Samot | So the value is being set. |
| 15:29.23 | Samot | What's the issue? |
| 15:30.43 | Marquel | the other extension doesn't get it. it's empty when trying to do ChannelRedirect(${EXTERNALCHANNEL}) in another context/channel: -- Executing [*666@phones:1] ChannelRedirect("SIP/7111-0000016c", ",std-inbound,i,1") in new stack |
| 15:30.59 | Marquel | should be right in front of ',std-inbound' but it isn't there. |
| 15:31.24 | Samot | OK.. |
| 15:31.41 | Marquel | exactly. |
| 15:31.45 | Samot | So whatever the global var value is.. |
| 15:31.58 | Samot | When set in [globals] is the value for the WHOLE call... |
| 15:32.27 | Samot | You need to do __EXTERNALCHANNL to make your overridden value apply beyond the channel you are on. |
| 15:32.37 | Samot | You need to do __EXTERNALCHANNEL to make your overridden value apply beyond the channel you are on. |
| 15:33.54 | Samot | Channel B has no idea what the values of vars on Channel A are. |
| 15:34.10 | Samot | Unless you make the vars inheritable. |
| 15:34.31 | Samot | Otherwise Channel B will load the [globals] vars with the values set in [global] |
| 15:34.36 | Marquel | doesn't help. |
| 15:35.01 | Marquel | double underscores in front and still empty. |
| 15:35.06 | Samot | 11:21:00 AM S<Samot> Marquel: Show the dialplan |
| 15:35.34 | Marquel | exten => *666,1,ChannelRedirect(${EXTERNALCHANNEL},std-inbound,i,1) |
| 15:35.42 | Samot | sigh |
| 15:35.46 | Samot | ALL OF IT |
| 15:35.56 | Marquel | that IS all of it. |
| 15:36.04 | Samot | That can't be all of your dialplan |
| 15:36.19 | Samot | Where's the [globals] section |
| 15:36.32 | Samot | where's the std-inbound section? |
| 15:36.42 | Samot | Why are you only showing pieces? |
| 15:36.52 | Samot | You are asking for help on finding where this issue is... |
| 15:36.59 | Samot | But you only show a part of the puzzle. |
| 15:37.23 | avb | __ is getting lost if youare using Local channels |
| 15:37.44 | avb | and in general they are working like crap |
| 15:37.46 | Marquel | the dialplan is 390 lines and i will need to redact it carefully. |
| 15:37.51 | Samot | FFS. |
| 15:37.53 | Samot | Nevermind. |
| 15:37.53 | avb | check out MASTR_CHANNEL |
| 15:37.57 | avb | check out MASTER_CHANNEL |
| 15:38.02 | avb | this is working somehow :) |
| 15:38.21 | Marquel | i am asking for one simple thing - i want a value to be available to whoever does a call in whichever context, whichever extension it might be. |
| 15:38.37 | Samot | OK |
| 15:38.47 | Samot | Thats channel inheritance. |
| 15:38.59 | avb | but i rewrote a solution not to utilize __ variables and storing all need stuff in the database |
| 15:39.02 | Marquel | i will use a fucking db entry if "global variable" is not what a "global variable" is everywhere else in programming but i really don't see where the entire 390 lines will help. |
| 15:39.04 | Samot | Or you need to use a GoSub or Macro on the Dial to set those vars. |
| 15:39.24 | Samot | I can't see how your settings VARS |
| 15:39.30 | Samot | I can't see if something isMESSING with them |
| 15:39.52 | Samot | I can't see if you had some WRONG in your dialplan code at any point in your logic. |
| 15:39.59 | Marquel | [globals] did contain "EXTERNALCHANNEL=" (literally) but does not do so anymore. |
| 15:40.05 | Samot | OK. |
| 15:40.14 | Samot | Dude, I've asked for specific things. |
| 15:40.19 | Samot | You don't want to give them. Fine. |
| 15:40.28 | avb | _CALLER=01796775267 will get cleared after you will exit from std-inbound |
| 15:40.50 | Marquel | db entry then. |
| 15:40.53 | Samot | Only the first child channel will inherit CALLER |
| 15:41.09 | Samot | Any channels that channel creates will not have CALLER |
| 15:41.16 | avb | right |
| 15:41.49 | avb | Marquel: listen to Samot , he is a smart guy :) |
| 15:46.09 | Marquel | CALLER is of no interest. |
| 15:46.17 | Marquel | i want CHANNEL to be available. |
| 15:47.22 | Samot | ${CHANNEL} is on every channel. |
| 15:47.28 | Marquel | Samot: right. |
| 15:47.40 | Samot | You're setting the value on the channel.. |
| 15:47.47 | Samot | You're just not passing that value to the NEXT channel. |
| 15:47.51 | Marquel | but i want the CHANNEL from std-inbound to be available in another context. |
| 15:48.01 | Samot | You have an inheritance issue. |
| 15:49.06 | Marquel | that might indeed be the case, yes. |
| 15:49.23 | Samot | It's not *might* |
| 15:49.28 | Samot | It *is the case. |
| 15:51.33 | Marquel | the database has solved it. |
| 15:51.52 | Samot | No, it didn't. |
| 15:51.59 | Samot | It just does it a different way. |
| 15:52.18 | Marquel | putting the ${CHANNEL} into the database did the trick. |
| 15:52.21 | Samot | So yeah, your over all issue of having that var on other channels...fixed. |
| 15:52.31 | Samot | Why you are not getting inheritance properly, not fixed. |
| 15:52.49 | Samot | So you didn't actually fix what you were doing. |
| 15:52.55 | Samot | You changed how you were doing it. |
| 15:52.59 | Marquel | as i said above: i expect a "global" to be available throughout all contexts at all times with the value it was last set to, but this is not the case with asterisk. |
| 15:53.07 | Samot | IT IS |
| 15:53.15 | Samot | If I set VARNAME= |
| 15:53.20 | Samot | That's it, in [globals] |
| 15:53.29 | Marquel | that's right. |
| 15:53.35 | Samot | The ${VARNAME} is loaded on every channel EMPTY |
| 15:53.36 | Marquel | that works here that way too. |
| 15:53.51 | Samot | If I set a NEW value to VARNAME in a context.. |
| 15:54.04 | Samot | It applies ONLY to the channel I am ON |
| 15:54.22 | Samot | If that channel creates a child, my updated value has to be INHERITED |
| 15:54.22 | Marquel | it did not work with VARNAME= in [globals]. |
| 15:54.33 | Marquel | yeah, but i do not create a child. |
| 15:54.48 | Samot | When you do your Dial() <-- That's the child. |
| 15:54.52 | Marquel | i am talking about an entirely new call, in a different context. |
| 15:55.25 | Marquel | yes, i'm not interested in that Dial() anymore. I want to abort it. from another call in a different context, even unrelated extension. |
| 15:56.55 | Marquel | hence the ChannelRedirect() from before (as discussed with pawiecki). |
| 16:06.56 | Marquel | thanks for your patience with me and the helpful insights all of you gave me. |
| 16:09.38 | *** join/#asterisk miralin (~Thunderbi@91.237.94.6) |
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| 18:20.38 | hdon | hi all :) anyone know of any free radio stations or chat rooms that can be listened to / participated in using asterisk? |
| 18:21.02 | Samot | You mean besides this one? |
| 18:21.33 | hdon | Samot, i didn't know there was a chan_irc module :3 |
| 18:21.53 | Samot | OK, so you want to integrate them into Asterisk. |
| 18:22.01 | hdon | but i'm specifically interested in something with SIP + RTP audio |
| 18:22.13 | hdon | yes :) |
| 18:22.19 | Samot | In what way? |
| 18:22.19 | hdon | it doesn't have to be ABOUT asterisk |
| 18:22.41 | hdon | idk like an internet radio station that you can use SIP+RTP to listen to instead of the more usual Internet radio technologies |
| 18:22.43 | Samot | Do you want Asterisk to use these things? |
| 18:22.54 | Samot | Or do you just want to bundle them with Asterisk? |
| 18:23.15 | Samot | So you want to stream music on hold to someone |
| 18:23.22 | hdon | no no |
| 18:23.25 | hdon | this is not a practical application\ |
| 18:23.30 | hdon | i just thought it would be something fun to do with asterisk |
| 18:23.43 | Samot | How would I listen to this as an end user? |
| 18:24.13 | hdon | with a softphone maybe |
| 18:24.19 | hdon | something like a party line would be perfect |
| 18:24.21 | Samot | Why? |
| 18:24.33 | hdon | mostly for fun |
| 18:24.51 | Samot | So no practical application. |
| 18:24.59 | Samot | Nothing a radio station would invest in. |
| 18:25.04 | Samot | Something for a podcaster. |
| 18:25.20 | hdon | yeah |
| 18:25.23 | hdon | something like that |
| 18:25.34 | Samot | You're basically talking about a conference bridge. |
| 18:25.54 | Samot | Where the admin (DJ/whatever) is the only one that can "talk" i.e. stream audio |
| 18:25.59 | Samot | Everyone is muted. |
| 18:27.13 | hdon | yeah that would work |
| 18:27.16 | hdon | except open to the public |
| 18:27.24 | Samot | OK. |
| 18:27.34 | Samot | As far as the chat goes.. SIP SIMPLE.. |
| 18:27.56 | hdon | well i know how to build it. i want to know if any already exist :) |
| 18:27.59 | Samot | But now you're talking all the users have to have a SIP account. |
| 18:28.03 | Samot | No. |
| 18:28.12 | Samot | Because there's no real world application for it. |
| 18:28.23 | Samot | I mean, it could exist. In someone's basement. |
| 18:28.47 | hdon | well, that someone's basement is somewhere... i'm gonna try to find it |
| 18:28.49 | Samot | For funsies that their friends or small social group know about. |
| 18:29.22 | Samot | Because really Asterisk isn't the right platform for just streaming audio and chatting in an app. |
| 18:29.24 | [TK]D-Fender | you'll find them next to the wild geese. |
| 18:29.36 | hdon | [TK]D-Fender, hopefully |
| 18:29.55 | Samot | You're trying to use telephony for web media apps. |
| 18:30.00 | Samot | Totally wrong way. |
| 18:30.04 | hdon | meh |
| 18:30.20 | hdon | i just want to have some fun sometimes |
| 18:31.03 | [TK]D-Fender | And picking the least practical starting points |
| 18:32.00 | hdon | that's part of the fun |
| 18:32.48 | Samot | You're the guy in the group that gets on everyone's case because playing chicken with cars on the freeway isn't fun. |
| 18:32.54 | Samot | "You could get killed" |
| 18:32.58 | Samot | "That's part of the fun" |
| 18:33.23 | Samot | "Come on guys, where is your sense of adventure. Screw logic!" |
| 18:37.10 | *** join/#asterisk Penguin (~xwQ5kwYl6@2001:470:1f07:26d:20c:29ff:fe62:be33) |
| 18:40.38 | *** join/#asterisk corretico (~corretico@200.91.143.34) |
| 18:42.14 | hdon | Samot, that doesn't seem fair |
| 18:42.34 | Samot | No, but it was kinda funny. |
| 18:42.38 | hdon | :) |
| 18:42.52 | Samot | And that's part of the fun! |
| 18:42.55 | hdon | :3 |
| 18:43.02 | *** join/#asterisk Mr_Pleb_Mgoo (~jakeb@103.46.213.148) |
| 18:44.22 | Samot | I mean there is a chance someone has done this before. |
| 18:44.26 | Samot | I don't know. |
| 18:44.50 | Samot | There is also a chance that someone tried it, it didn't work so that's why there's nothing out there on it. |
| 19:28.41 | startledmarmot | hdon: I heard, a while back, about an open community of Asterisk-based conference rooms dedicated to certain diaspora communities |
| 19:28.46 | *** join/#asterisk areski (~areski@37.223.245.20) |
| 19:29.04 | startledmarmot | essentially a place for people the Hmong to hang out and speak Hmong all day |
| 19:29.15 | startledmarmot | because it's so infrequently heard in real life |
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| 21:16.37 | jamesaxl | I am connecting With appendixes with openvpn. should I use install asterisk on every appendix or one is enough ? |
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| 21:50.45 | Samot | What is an "appendix"? |
| 21:50.53 | Samot | What happens if it bursts? |
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| 22:18.38 | *** join/#asterisk coch (500c3b45@gateway/web/freenode/ip.80.12.59.69) |
| 22:18.47 | coch | Hi |
| 22:19.00 | coch | Is there anybody ? |
| 22:19.35 | coch | I'm trying to configure an asterisk server |
| 22:20.14 | fauxalliance | and... |
| 22:20.20 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 22:20.42 | coch | And I Have the following error : chan_sip 4274 serious network trouble |
| 22:20.52 | fauxalliance | coch: what's the issue, qu'est qui ce passe |
| 22:21.37 | coch | Je peux passer en francais si tu préfères |
| 22:21.38 | fauxalliance | coch: get some sip debug and pastebin the logssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssss |
| 22:22.01 | coch | Okay c'est parti je t'envoie la stacktrace |
| 22:22.20 | fauxalliance | i craked my touchscreen, and ubuntu and xinput needs an angry bug report |
| 22:24.36 | fauxalliance | coch: anglais préféré, you'll get more help |
| 22:24.46 | fauxalliance | in french, less sarcasm |
| 22:24.48 | UncleKiwi | hi, I am wanting to configure 'wake up calls' for a hotel I have installed asterisk into i am googling but I thought I would ask here if anyone can suggest a link to the best path to follow |
| 22:24.57 | coch | I didn't forgot you, I'm trying to send you the stack trace |
| 22:25.04 | fauxalliance | sip debug please |
| 22:25.26 | coch | How can I have the SIP debug ? |
| 22:25.32 | coch | In the output ? |
| 22:25.45 | Samot | coch: Stop. |
| 22:26.07 | Samot | coch: That error 99.9% of the time means your configuration is not correct. |
| 22:26.33 | coch | Yes thanks Samot, I know, but I didn't manage to put the good conf |
| 22:26.39 | Samot | OK. |
| 22:26.45 | coch | I'm working on it for 4 hours |
| 22:26.51 | Samot | So looking at a SIP debug isn't going to fix your config. |
| 22:26.56 | Samot | Are you using Chan_SIP? |
| 22:27.00 | fauxalliance | asterisk -rvvvvvv |
| 22:27.09 | fauxalliance | sip set debug on |
| 22:27.13 | fauxalliance | or something somthing |
| 22:27.14 | Samot | Or are you using PJSIP? |
| 22:27.19 | fauxalliance | ^^ |
| 22:27.25 | coch | Samot : PJSIP |
| 22:27.28 | Samot | OK. |
| 22:27.31 | Samot | ~pb |
| 22:27.31 | infobot | well, 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. |
| 22:27.37 | Samot | Show your pjsip.conf file. |
| 22:27.37 | fauxalliance | maudit tabarnac |
| 22:27.43 | coch | @fauxalliance, okay I'm writing those commands |
| 22:27.47 | Samot | No. |
| 22:27.48 | fauxalliance | stp |
| 22:27.48 | Samot | Stop. |
| 22:27.52 | Samot | Show your config. |
| 22:28.05 | coch | Okay I send you the conf pjsip file |
| 22:28.13 | Samot | Pastebin |
| 22:28.14 | Samot | ~pb |
| 22:28.15 | infobot | pastebin is, like, 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. |
| 22:28.25 | coch | Ok, I put it on pastepin |
| 22:28.26 | fauxalliance | Samot: thx1138 |
| 22:28.29 | Samot | Put the link in the channel. |
| 22:28.35 | fauxalliance | coch: STP |
| 22:28.39 | coch | Okay Samot |
| 22:28.45 | coch | Yes fauxalliance ? |
| 22:28.57 | fauxalliance | PLEASE |
| 22:29.16 | fauxalliance | SIL TU PLAIT |
| 22:29.29 | coch | Yes tell me fauxalliance ? |
| 22:29.34 | fauxalliance | paste the bloody logs |
| 22:29.38 | fauxalliance | in a pastebit |
| 22:29.41 | fauxalliance | s/t/m |
| 22:29.42 | fauxalliance | n |
| 22:30.25 | coch | Okay I'll do it too |
| 22:32.07 | coch | https://pastebin.com/Xr3UC68r |
| 22:32.12 | coch | First one : the conf file |
| 22:32.39 | coch | https://pastebin.com/11chfCaA |
| 22:32.47 | coch | Second one : debug log |
| 22:33.21 | fauxalliance | thats easy, Samot, take it away. |
| 22:34.09 | fauxalliance | coch: quel version de Asterisk? DHADI? |
| 22:34.47 | coch | ASterisk : 13.17.2 |
| 22:34.52 | fauxalliance | vraiment |
| 22:35.03 | coch | Yes, why ...? |
| 22:35.31 | coch | DAHDI : 2.10.2 |
| 22:35.35 | fauxalliance | ok ok |
| 22:36.20 | fauxalliance | there are multiple issues, you might need to enable the dhadi debug so we can see whats going on with the caller id error. |
| 22:36.40 | coch | OKay, how I can activate it ? |
| 22:37.20 | Samot | Uhm. |
| 22:37.25 | Samot | You said you're using PJSIP |
| 22:37.33 | coch | Yes |
| 22:37.33 | Samot | And all I see is DAHDI and Chan_SIP |
| 22:37.58 | fauxalliance | ^^ |
| 22:38.04 | coch | The conf file I paste you is the pjsip.conf file |
| 22:38.09 | Samot | OK |
| 22:38.14 | Samot | Because you said you're using it. |
| 22:38.27 | Samot | Your debug has no one thing related to PJSIP in it. |
| 22:38.45 | Samot | *not |
| 22:39.34 | coch | I send you and other trace where you will see pjsip error too |
| 22:40.06 | fauxalliance | core set debug on |
| 22:40.22 | coch | 'core set debug on' is done |
| 22:41.04 | fauxalliance | bring bring... où est-ce que ça fait mal |
| 22:41.18 | fauxalliance | paste all the log please |
| 22:41.53 | coch | https://pastebin.com/NJZ2YhAz |
| 22:42.51 | fauxalliance | dix lignes... patrout!??! |
| 22:42.57 | fauxalliance | all of it |
| 22:43.17 | fauxalliance | slurs in mulptiple languages |
| 22:43.20 | coch | It's all the log I have |
| 22:43.36 | coch | When I make a phone call I have that stack trace |
| 22:44.17 | fauxalliance | "what rolls downstais, alone or in pairs, and over your neighbours dog..." it's log, it's better than bad, it's good. |
| 22:44.42 | *** join/#asterisk rwb (~Thunderbi@65.183.151.121) |
| 22:44.49 | fauxalliance | coredump? |
| 22:45.24 | coch | coredump in asterisk ? |
| 22:46.35 | *** part/#asterisk kharwell (kharwell@nat/digium/x-kixjbwlhkmpnxiby) |
| 22:46.35 | fauxalliance | â[20:13] â<âcochâ>â When I make a phone call I have that stack trace |
| 22:46.43 | fauxalliance | ^^ show it please |
| 22:46.57 | [TK]D-Fender | Chan_sip isn't loaded |
| 22:46.58 | fauxalliance | it's bigger than ten lines, i pray |
| 22:47.05 | coch | https://pastebin.com/FYXp6c9P |
| 22:47.30 | [TK]D-Fender | <PROTECTED> |
| 22:47.31 | coch | Line 1Ã |
| 22:47.33 | coch | 10 |
| 22:47.38 | [TK]D-Fender | this also looks wrong. |
| 22:47.49 | fauxalliance | parlant du diable |
| 22:47.50 | [TK]D-Fender | You shouldn't have a peer & IP there at the same time |
| 22:47.56 | coch | The warning appears when I make a phone call |
| 22:48.11 | [TK]D-Fender | <[TK]D-Fender> Chan_sip isn't loaded <------ |
| 22:48.13 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 22:48.14 | fauxalliance | it explodes |
| 22:48.16 | fauxalliance | !!!!! |
| 22:48.56 | coch | Okay, maybe we can start to fix the bugs one by one....... =D |
| 22:49.11 | [TK]D-Fender | Fix your chan_SIP so it loads |
| 22:49.21 | coch | Okay, how can I do that ? |
| 22:49.30 | [TK]D-Fender | LOOK AT IT |
| 22:49.32 | fauxalliance | module load sip shit |
| 22:49.43 | fauxalliance | core load sip shit even. |
| 22:50.12 | [TK]D-Fender | module load chan_sip.so |
| 22:50.14 | [TK]D-Fender | ^ |
| 22:50.25 | [TK]D-Fender | and look at your config when you see the result of that |
| 22:50.35 | [TK]D-Fender | already suspects the usual SNAFU |
| 22:50.53 | [TK]D-Fender | <coch> And I Have the following error : chan_sip 4274 serious network trouble <--- which this is indicative of |
| 22:50.54 | coch | Okay I'll start to do that : module load chan_sip.so |
| 22:52.14 | coch | module load chan_sip.so -> I send you the trace |
| 22:52.49 | coch | https://pastebin.com/8TdTXEax |
| 22:53.18 | coch | Okay now when I make a phone call I have this stacktrace : |
| 22:53.54 | [TK]D-Fender | [Oct 14 00:51:53] WARNING[2983]: chan_sip.c:32980 reload_config: Failed to bind to 0.0.0.0:5060: Address already in use |
| 22:53.55 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 22:53.59 | [TK]D-Fender | Eaxctly as suspected |
| 22:54.03 | coch | https://pastebin.com/E8247Bvw |
| 22:54.06 | [TK]D-Fender | PJSIP is configured to use the SAME port |
| 22:54.10 | [TK]D-Fender | and it is FUCKING YOU |
| 22:54.17 | [TK]D-Fender | You can't have them fight over the same port |
| 22:54.27 | [TK]D-Fender | So pick one and make it DIFFERENT than the other |
| 22:54.28 | coch | Okay, how can I fix that ? |
| 22:55.19 | coch | Okay can I remove pjsip or I can the port ? |
| 22:55.29 | coch | *change the port |
| 22:56.16 | [TK]D-Fender | either |
| 22:56.35 | [TK]D-Fender | Make them stop fighting over the same port |
| 22:56.37 | [TK]D-Fender | Doesn't matter how. |
| 22:56.44 | [TK]D-Fender | Confiure your system accordingly |
| 22:57.13 | coch | Okay, how can I change the port of the pjsip ? On the bind ? |
| 22:57.20 | fauxalliance | configure/confuse |
| 22:57.24 | fauxalliance | SAME THING |
| 22:58.23 | coch | @fauxalliance : I don't understand |
| 23:01.06 | [TK]D-Fender | its in your Transport |
| 23:01.13 | [TK]D-Fender | Read the sample config |
| 23:01.59 | coch | bind=0.0.0.0:5061 |
| 23:02.13 | coch | Okay thank, I'm changing it. And after I reload the sip |
| 23:03.59 | coch | Done |
| 23:04.47 | Samot | Need to restart asterisk |
| 23:04.51 | Samot | After port changes |
| 23:04.54 | coch | https://pastebin.com/eST1piqr |
| 23:04.56 | Samot | Not just reload sip |
| 23:05.30 | coch | asterisk restarted |
| 23:07.42 | coch | Holly chit |
| 23:07.49 | coch | It's fucking working |
| 23:07.53 | coch | Thanks a lot men |
| 23:07.55 | Samot | Yeah |
| 23:08.25 | Samot | 6:26:07 PM <Samot> coch: That error 99.9% of the time means your configuration is not correct. |
| 23:08.36 | Samot | Called it 40 mins ago |
| 23:09.10 | coch | Yeah but I didn't realize at this time |
| 23:09.18 | coch | I'm a beginner on this software |
| 23:09.31 | coch | Thanks for your time @all |
