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| 01:12.53 | *** join/#asterisk Lope (~Lope@192.228.187.122) |
| 01:15.34 | Lope | I'm recording the bridged part of a call betwenn the caller and the member of a queue. How can I add/modify/rename the MONITOR_FILENAME to include the queue member's username? |
| 01:16.23 | Lope | Currently the filename I'm saving is storing the caller's ID and number etc. But I also want to record what queue member they spoke to? |
| 01:18.14 | Lope | there does not seem to be any QUEUE_MEMBER_USERNAME sort of environment variable. There's one called QUEUEMEMBER but it only contained ''. |
| 01:18.26 | Lope | Perhaps I need to name my queue members... |
| 01:29.15 | Lope | Okay I gave my queue members names, but QUEUEMEMBER channel var still contained '' |
| 01:31.43 | Lope | I've looked around at the various QUEUE_ functions and don't see any way to get the member's name that they spoke to? |
| 01:35.41 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 01:53.48 | Lope | How do you write code for asterisk to run on an event? |
| 01:54.10 | Lope | such as ManagerEvent_AgentConnect ? |
| 02:05.13 | [TK]D-Fender | Asterisk doesn't run on that |
| 02:05.21 | [TK]D-Fender | that is AMI and it's well documented |
| 02:10.14 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 02:11.32 | Lope | Okay. Mixmonitor is currently not recording my calls. In queues.conf I've got monitor-type=MixMonitor monitor-format=gsm |
| 02:12.46 | Lope | I'm looking at the Queue application's options, and I see there are 4 options relating to enabling MixMonitor to record the call to disk. but they're all say that either the caller or the member can record the call... these options are w W x X ... I just want the call to be recorded automatically? |
| 02:15.13 | Lope | Before starting the queue I have same => n,Set(MONITOR_OPTIONS=ab) same => n,Set(MONITOR_FILENAME=${UNIQUEID}.gsm) |
| 02:15.47 | Lope | Do I need to run the mixmonitor app? (before the queue app) Because currently I'm not. |
| 02:16.50 | [TK]D-Fender | younever specify the extension on the file like that |
| 02:17.21 | [TK]D-Fender | that's what monitor-format is for |
| 02:17.54 | [TK]D-Fender | Show the actual full config and the actual call (with proper debug) and your actual monitor folder |
| 02:22.18 | Lope | okay one moment please. (btw I removed the file extension and it's still not recording) |
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| 02:28.38 | Lope | extensions.conf and queues.conf http://codepad.org/Krkr82yc |
| 02:30.49 | Lope | ls -la /var/spool/asterisk http://codepad.org/19ABMAta |
| 02:32.08 | [TK]D-Fender | drwxr-x--- 2 asterisk asterisk 4096 Jan 27 04:23 monitor |
| 02:32.26 | Lope | ot |
| 02:32.30 | Lope | it's writable? |
| 02:32.52 | Lope | writable to asterisk owner? |
| 02:32.54 | [TK]D-Fender | THE CONTENTS OF THE FOLDER |
| 02:32.59 | [TK]D-Fender | Files don['t go BELOW that |
| 02:33.47 | Lope | empty: http://codepad.org/AX8PaBkM |
| 02:34.15 | [TK]D-Fender | Good. Now you've at least checked the actual place they should gett saved. |
| 02:34.21 | [TK]D-Fender | Then on to the more obvious.... |
| 02:34.31 | [TK]D-Fender | You should re-read the sample queues.conf and realize you screwed where you put your parameters <--- |
| 02:35.58 | Lope | Thanks :) okay it's working now. I moved the mixmonitor stuff to the actual queue config. |
| 02:36.26 | [TK]D-Fender | VERY FINE MANUALS |
| 02:36.28 | Lope | I'll do a more thorough check of the queues.conf for the rest now. |
| 02:36.40 | [TK]D-Fender | starts a new hash-tag |
| 02:37.04 | Lope | Thanks for your help :) |
| 02:37.07 | [TK]D-Fender | #veryfinemanuals |
| 02:37.15 | [TK]D-Fender | sits back and waits for it to start trending |
| 02:38.12 | Lope | Lastly, I found a way to rename my recordings such that I can note the member name that the caller spoke to, in the recorded file. By storing the UUID in the filename, then parsing the queue log file. |
| 02:38.35 | Lope | It seems to be the most practical method, would you agree? |
| 02:39.29 | Lope | Thus far I've not found a way to access the member username or name that the caller spoke to, inside extensions.conf via any channel variables or functions. |
| 02:42.32 | [TK]D-Fender | As there is none |
| 02:42.50 | [TK]D-Fender | all you have are the channel names directly |
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| 02:43.10 | [TK]D-Fender | You'll have to look it up AFTER the call and clean up after it |
| 02:55.48 | Lope | Cool, thanks :) |
| 02:56.09 | Lope | Much appreciated! |
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| 04:28.00 | Lope | What about this, in queues.conf: UpdateCDR "...populating CDR dstchannel field of a call with an agent name, which you can set at the login time with AddQueueMember membername parameter" |
| 04:31.46 | [TK]D-Fender | interesting. |
| 04:31.55 | [TK]D-Fender | Depends when you can get access to that. |
| 04:38.22 | Lope | Ah I figured it out/found a solution. There's an option called setinterfacevar inside queues.conf |
| 04:38.41 | Lope | when that is yes, it sets MEMBERNAME channel var |
| 04:46.36 | *** join/#asterisk iamb (~iamb@unaffiliated/iamb) |
| 04:49.39 | [TK]D-Fender | And does it turn out like you'd hope? |
| 04:49.46 | [TK]D-Fender | is it set in time to USE? |
| 04:51.56 | Lope | I'm still figuring that part out... |
| 04:52.06 | Lope | right now I'm finding that MONITOR_FILENAME is not being used. |
| 04:52.32 | Lope | Regardless of what I set MONITOR_FILENAME to before running the Queue app, the filename is just the UUID :/ |
| 04:52.38 | [TK]D-Fender | Show it |
| 04:52.49 | Lope | okay 1 sec |
| 04:53.26 | *** join/#asterisk O47m341 (~Suzeanne@75-103-145-152.ccrtc.com) |
| 04:54.32 | Lope | ah, must be a problem in my syntax somewhere. It's working now.. |
| 05:07.02 | Lope | lol, my goto was pasing where I was setting the channel var haha |
| 05:08.37 | *** join/#asterisk iamb (~iamb@unaffiliated/iamb) |
| 05:19.40 | Lope | I got it working... same => n,System(mv /var/spool/asterisk/monitor/${MONITOR_FILENAME}.gsm /tmp/${MEMBERNAME}-${MONITOR_FILENAME}.gsm) |
| 05:37.11 | Lope | lastly, it might be a bit ambitious but it would be cool to get the length of the call in seconds. So far I'm not seeing any channel var etc. |
| 05:38.39 | *** join/#asterisk O47m341 (~Suzeanne@75-103-145-152.ccrtc.com) |
| 05:39.05 | [TK]D-Fender | just "getting" it doesn't sound interesting yet. |
| 05:39.13 | [TK]D-Fender | What do you intend to DO with this fact? |
| 05:40.06 | Lope | Store it in the filename? :) |
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| 05:40.34 | Lope | so foo-100.gsm (I know the call was 100 seconds long) |
| 05:40.44 | Lope | It's not critical by any means. |
| 05:43.24 | Lope | What's better in terms of quality and small size (I don't care about CPU usage to compress) gsm or ogg? |
| 05:45.30 | [TK]D-Fender | You should stop assuming everything is a channel variable. |
| 05:45.35 | [TK]D-Fender | VERY little of anything is |
| 05:45.42 | [TK]D-Fender | and DUmpChan will give you all of them |
| 05:45.54 | [TK]D-Fender | So you actually never have to ask this |
| 05:46.19 | [TK]D-Fender | Then you can think ahead just a tiny bit and script getting that answer from an outside script |
| 05:47.48 | Lope | Thank you, that will be very useful |
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| 07:09.49 | Lope | I'm reading a digit into a channel variable, using Read. If the user doesn't enter a value and it times out, I want to overwrite it with my own default value. I don't see a way to set a default value for Read. What's the best way to do it? ExecIf? |
| 07:18.58 | *** join/#asterisk rogger_h (~rogger_h@bzq-79-180-97-72.red.bezeqint.net) |
| 07:19.02 | rogger_h | Hello all :) |
| 07:19.43 | rogger_h | anyone is arround? |
| 07:20.01 | Lope | Can I use a function inside a Set application? |
| 07:21.24 | Lope | something like: same => n,Set(blah=foo-STRREPLACE(CallerIDName,' ','_')) |
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| 07:29.19 | rogger_h | im new to asterisk and i have a question, i have 5 representive postisions in my office and i want them to get the phone calls using the pc, i just didnt get how do i do that with asterisk .. |
| 07:31.16 | Lope | hi rogger_h your question is extremely vague, if you can describe what you need more specifically then maybe someone can help you? |
| 07:32.40 | Lope | I'm trying to figure out how to concatenate the STRREPLACE variable with a string inside a Set() app. So far no dice. I've tried foo-:STRREPLACE foo-$STRREPLACE foo-${STRREPLACE etc |
| 07:34.38 | Lope | Even a really simple case doesn't work. It's storing the literal characters in the variable: same => n,Set(CallerIDNameU=STRREPLACE(CallerIDName,' ','_')) |
| 07:35.18 | rogger_h | i have a web app that control orders and invoices and stuff, that webapp is available in each computer of my employees. i want to connect asterisk into the computers so when incoming call is done a popup will just inside my webapp and by click" answer" the call will come trough the computer speakers and mic so they can talk. |
| 07:38.23 | Lope | Oh I see I'm getting Channel Info mixed up with Channel Variables. When I run DumpChan it first displays stuff under an "Info" heading, one of those is CallerIDName. |
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| 07:40.03 | Lope | rogger_h: so basically you want to create your own custom softphone, with the constraint that it must work in a browser. That might be very challenging. |
| 07:40.18 | Lope | rogger_h: check out webRTC |
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| 07:42.09 | rogger_h | shit, u used the word "challenging" thats means im going to die lol |
| 07:42.19 | rogger_h | i atually need a quick solution |
| 07:42.41 | Lope | quick solution: install softphones that already exist on each PC. |
| 07:43.12 | rogger_h | but it must a pear on the web browser |
| 07:43.15 | Lope | Not knowing how to use Asterisk means you will not have a quick solution very easily. It's quite a versatile esoteric piece of software. |
| 07:43.27 | *** join/#asterisk areski (~areski@80.174.128.27.dyn.user.ono.com) |
| 07:43.38 | Lope | Then you probably won't have a quick solution. |
| 07:43.52 | rogger_h | like "There is an incoming call from "01902319230" click <button> to accept " |
| 07:44.06 | rogger_h | Yes, thats why i was comming here |
| 07:44.12 | Lope | Like I said, google webRTC, that's about all I can help you with. |
| 07:44.27 | rogger_h | ok |
| 07:44.29 | rogger_h | i will check |
| 07:45.27 | Lope | VOIP is very complicated. Don't underestimate it like I did. You'll be snowed under. |
| 07:45.46 | Lope | If you want easy, use google hangouts ;) |
| 07:50.44 | rogger_h | the thing is that it MUST communicate with my webapp |
| 07:50.46 | Lope | How can I access these channel info items, for example CallerIDName ? |
| 07:51.31 | Lope | rogger_h: if you want to hack something and use a normal softphone, you can implement a transparent SIP proxy and sniff what's going on. |
| 07:52.33 | Lope | rogger_h: you can't just expect VOIP to work in a web-app. SIP is a very fragile/high-maintenance protocol. It won't work in a lot of situations where a web-app will work. |
| 07:53.07 | rogger_h | wait wait i dont need the VOIP to work in webapp, BUT i do ned to get the data and control the call fromt he webapp |
| 07:53.32 | rogger_h | but if i click on "answer" on teh webapp it will send a command to the call to asnwer |
| 07:53.49 | Lope | You can't do that without hacking up your own custom softphone or using webRTC |
| 07:54.19 | Lope | And if you think you're finding this difficult, you're in for a world of trouble when your users try to get it working on their networks. |
| 07:54.31 | Lope | (Just being honest/realistic) |
| 07:54.42 | rogger_h | OK what about REST , my webapp using WEBSOCKET so i can use my server to send a command to browser and when i click on that it will send rest command to my server |
| 07:54.45 | rogger_h | is that better? |
| 07:55.10 | rogger_h | so the communication wont get directly from teh webbrowser |
| 07:55.18 | Lope | rogger_h: it doesn't address the real issue at all. |
| 07:55.31 | rogger_h | Where are u from Lope? |
| 07:56.01 | Lope | I'm not here to have personal chats. I'm working. I get help on #asterisk, so I give help when I can. |
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| 07:56.52 | rogger_h | Yes but i need some help and i dont mind PAY for understanding teh concept, so i was asking maybe u can give me few mins on skype call and i will pay u for that , i have a meeting in 1 hour and i want to present the options |
| 07:56.57 | rogger_h | just tell me how much u want |
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| 07:59.57 | Lope | I don't know more than the basic concepts involved with WebRTC, so I'm not qualified to give advice on it. I only have experience with Asterisk, SIP protocol and softphones. |
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| 08:04.58 | Lope | What is "Information on channel" They're not channel variables, how can I access them from my extensions.conf? |
| 08:04.59 | rogger_h | there is noones here and my meeting is in an hour so i rather talk to u |
| 08:05.02 | rogger_h | if u dont mind |
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| 08:25.07 | Lope | Oh I see, callerID is obtained from the callerid function. |
| 08:33.12 | Lope | I got my STRREPLACE working. I see it must be double quotes and can't take another function as an input. |
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| 08:51.07 | Zogot | Ahoy all |
| 08:59.46 | Lope | hi |
| 09:29.09 | Zogot | hmm, attempting to build asterisk from sources, i get, when ran make menuselect.makeopts : Unknown value '' found in build_tools/menuselect-deps for HAVE_ADDRESS_SANITIZER |
| 09:29.23 | Zogot | and 3 others, HAVE_LEAK_SANITIZER, HAVE_THREAD_SANITIZER HAVE_UNDEFINED_SANITIZER |
| 09:29.29 | Zogot | from asterisk-13-latest tar |
| 09:29.35 | Zogot | sorry, current tar |
| 09:30.02 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 09:31.18 | Zogot | oh, seems its a really old issue, someone also has a comment on the wiki about it, from Dec 2014 |
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| 10:27.29 | *** join/#asterisk ballock (~ballock@rohan.tokarscy.com) |
| 10:36.01 | ballock | I need some help, or at least direction on where to dig - I need to direct incoming phonecall at 2 SIP handsets, one of them is a SIP client on a mobile, and drop the incoming call if the mobile is away. |
| 10:40.15 | Lope | Zogot: I built Asterisk recently. I didn't have that issue. Make sure you start with a clean VM and follow the instructions exactly. |
| 10:42.02 | Lope | ballock: you'll need to add all of your sip endpoints in sip.conf then in extensions.conf you'll need to have the relevant contexts, and check the status of the endpoint and dial if it's online. |
| 10:46.52 | ballock | Lope: thanks, that should get me going |
| 10:47.46 | Lope | You can literally just have 1 line in your extensions.conf file. [mycontext] exten => 5555555555,1,Dial(SIP/foobar,30) ;timeout 30s |
| 10:48.05 | Lope | if your sip phone is online, it will get through, if it's not, it won't. |
| 10:48.43 | ballock | Lope: yeah, but I want both to ring if the mobile is online |
| 10:48.59 | Lope | both to ring and one to answer? |
| 10:49.07 | ballock | yes |
| 10:49.24 | Lope | so after one of the two picks up, you want the call connected to the one who picked up? |
| 10:49.39 | ballock | Lope: yes |
| 10:49.39 | Lope | Not both? |
| 10:50.20 | Lope | You can use a queue for that. |
| 10:52.04 | jkroon | ballock, Dial(SIP/a&SIP/b,30) |
| 10:52.24 | Lope | ah, jkroon schooling us. |
| 10:52.41 | jkroon | prefers Queues for more complex cases. |
| 10:53.39 | ballock | ok, sounds simpler |
| 10:53.41 | jkroon | someone will need to explain the more complex scenarios that's possible with PJSIP but as I understand it a single SIP peer can have multiple registered Contact: locations, whether that's of any use in this case is something I can only debate about. |
| 10:53.41 | Lope | jkroon: yeah I was hesitant to recommend a queue. I didn't know Dial can do that. I just looked at the help to confirm. It's massive! that was unexpected. |
| 10:54.19 | jkroon | Lope, I combine queue's with things like VM detection, so even forwarding to external numbers, and confirming that I don't get VM before actually connecting the caller. |
| 10:54.45 | Lope | ballock: chan_sip is the sortof obsolete module and pjsip is the new one. But "they" say pjsip is harder to use. I've not tried it myself. |
| 10:55.00 | Lope | VM? |
| 10:55.09 | jkroon | voicemail in this context. sorry about that. |
| 10:55.16 | Lope | ah, all good :) |
| 10:55.31 | Lope | jkroon: very clever :) |
| 10:56.31 | file | PJSIP has a more complete/expressive/SIP configuration which is what throws people off, so you can use templates or wizards (they consolidate it down to 1 entry like sip.conf) to help |
| 10:56.48 | *** join/#asterisk Zogot (~Adium@185.21.52.255) |
| 10:57.06 | jkroon | :). not particularly, the G option to Dial() IIRC and some special trickery in that context (remembering that if the called party hangs up before Read() or AMD() completes) you need to have a default case). |
| 10:57.20 | ballock | still, I need to use some GotoIf(...${DEVICE_STATE}=NOT_INUSE) before that, right? Otherwise both will call even if the mobile is away? |
| 10:57.33 | jkroon | ballock, no. |
| 10:57.55 | jkroon | just disable call-waiting on the devices themselves if that's an issue for you to ring something in the person's ear. |
| 10:59.30 | ballock | jkroon: It's not about that. It's a little bit more complicated, though the rest of the plan I have already figured out and working: |
| 11:00.23 | ballock | I have a weak GSM range at home, so I enabled MultiSim at my provider and bought a huawei and a dongle for chan_dongle. |
| 11:00.33 | ballock | This part works. |
| 11:00.38 | Lope | ballock: from what I understand you can just dial both, shouldn't be a problem if one is offline. |
| 11:01.01 | ballock | The provider does the multisim in a different way than I'd expect - |
| 11:01.34 | jkroon | ballock, oh wait. |
| 11:01.49 | ballock | When somebody calls the mobile number, the provider dials chan_dongle sim and only if it rejects, it rings the real mobile through GSM. |
| 11:01.56 | jkroon | you need to make what I call a "break out call" in order to ring the mobile - it doesn't register via SIP with you? |
| 11:02.49 | jkroon | ok ok, so you have that incoming channel, probably DONGLE/name-????? something. incoming is fine. |
| 11:02.55 | jkroon | what's the problem here? |
| 11:03.16 | ballock | So what I do is I allow asterisk to dial the stationary SIP phone for 4s, then reject it. |
| 11:03.29 | ballock | then the provider directs the call at the mobile. |
| 11:03.40 | jkroon | 4s is very short. |
| 11:03.47 | ballock | jkroon: exactly |
| 11:03.57 | jkroon | you have wifi? |
| 11:04.08 | jkroon | ie, your mobile is on your local LAN? |
| 11:04.20 | ballock | but when I'm on the move, the phone rings at home, and my mobile doesn't get the incoming call for the first 4s |
| 11:04.35 | jkroon | that you'll need to have the provider change I think. |
| 11:04.37 | ballock | so I can't extend it cause I won't know anyone's calling if I'm away |
| 11:04.37 | jkroon | if they can. |
| 11:05.03 | WIMPy | Why don't you swap the SIMs? |
| 11:05.03 | jkroon | or you need to flip the DND on the phone at home whenever you come in and leave. |
| 11:05.28 | jkroon | or that ... |
| 11:05.29 | ballock | jkroon: not really, multisim is really rare at the providers here, and they just set it one way |
| 11:06.01 | ballock | it's mostly setup for the car kits with sims, when you start the car, it receives the phonecall instead of your phone |
| 11:06.19 | jkroon | that makes sense. until your wife drives the car :p |
| 11:06.33 | ballock | jkroon: correct |
| 11:06.34 | jkroon | which is where I think the bluetooth solution works better. |
| 11:06.51 | jkroon | if phone is in car, register to it as a handsfree and use the same SIM. |
| 11:07.09 | jkroon | back to your problem - what about WIMPy's suggestion? |
| 11:07.15 | ballock | jkroon: yes, that's what you do nowadays |
| 11:07.32 | *** join/#asterisk degan6 (~degan6@c-73-151-71-161.hsd1.ca.comcast.net) |
| 11:07.46 | ballock | the dongle SIM does not receive SMSes nor can access the Internet |
| 11:08.24 | jkroon | ok, so if you swap them you won't have internet on your phone, nor will you get sms's. |
| 11:08.32 | Lope | at first I was going to record my calls in gsm format. But now I'm thinking record in wav16 then compress with ogg? |
| 11:09.04 | jkroon | Lope, isn't ogg optimized for music? |
| 11:09.05 | ballock | so the best I came up with is to install a SIP client on the mobile, and check its status, and reject the call promptly if the mobile's out, call long otherwise |
| 11:09.25 | Lope | jkroon: I suppose it is. But isn't that a good thing? it's good quality audio? |
| 11:10.02 | jkroon | Lope, shouldn't matter much either way. we currently record channels separately using g.729 and then post-process to end up with .g729, and then allow download in .mp3 (which is much, much larger) |
| 11:10.26 | Lope | oh, why do you record separately? |
| 11:10.28 | jkroon | Lope, odly enough, human speach compression and music seems to be two different beasts, but I'm not a specialist. |
| 11:10.32 | jkroon | g729 channels licenses. |
| 11:10.39 | WIMPy | ballock: Maybe you sould have take the new SIM with a new contract and us Forwarding instead. |
| 11:10.55 | jkroon | i can just framehook them in g729 format, without having to decode/encode during the call. |
| 11:11.18 | Lope | jkroon: oh yeah, cos you're working at massive scale? |
| 11:11.32 | ballock | WIMPy: afaik, when I use forwarding, I get my other mobile's phone number as the caller, which is inconvenient |
| 11:11.45 | Lope | I'd like to record in a format that I can listen to on my android phone. |
| 11:11.49 | jkroon | Lope, something like that. |
| 11:11.52 | Lope | I reckon ogg is a good choice for that. |
| 11:12.02 | jkroon | ogg is indeed a sensible choice. |
| 11:12.13 | Lope | gsm is a little crappy sounding. |
| 11:12.17 | WIMPy | ballock: You should get both. |
| 11:12.25 | Lope | g722 is probably a good one, but I don't know if I can play it on android. |
| 11:13.45 | ballock | WIMPy: but it's same problem - I need to tell if I should call or forward based on the mobile being home or not. |
| 11:13.47 | Lope | jkroon: I'm mostly interested in quality. not looking for the ultimate in space savings. I'll likely delete all calls every so often. |
| 11:13.55 | jkroon | WIMPy, ballock - that really depends on the way the forwarding works. |
| 11:14.03 | jkroon | Lope, then wav is the way to go :) |
| 11:14.07 | jkroon | HUGE though. |
| 11:14.17 | Lope | jkroon: haha yeah wav is too crazy. |
| 11:15.02 | jkroon | but it doesn't compress at all. any other lossless codecs? I'm not aware of any format. so for voice g729 seems to work well, as does speex, however, i doubt either of those will play natively on your android, so ogg is probably your best overall choice. |
| 11:15.59 | jkroon | WIMPy, ballock - if your mobile is set to forward and both phones is with the same provider it should (based on my limited knowledge) show both. any other way is unlikely to show both. at least that's my experience in ZA. |
| 11:17.30 | ballock | jkroon: that's a nice idea to evaluate, would dismiss the issue of using SIP client on the mobile... |
| 11:17.54 | ballock | but yeah, I still need to say if my phone is home or not |
| 11:18.03 | MajesticFudgie | How would I catch a 404 in Asterisk and play an invalid message |
| 11:18.03 | ballock | DND setting on the phone is not an option |
| 11:18.28 | MajesticFudgie | I'm trying exten => i,1,Answer() to answer invalid numbers but it doesnt do anything |
| 11:19.16 | ballock | first, no such button on the phone, second, the SIP phone uses two handsets internally through DECT, so I'd need to DND on both, third - it's too much hassle |
| 11:19.25 | jkroon | i gets executed when no other extensions matches, not when something bad happens. |
| 11:19.51 | MajesticFudgie | Thats the intention here |
| 11:20.01 | MajesticFudgie | but it simply doesnt and my sip client shows the 404 |
| 11:20.15 | Samot | jkroon, the i special extension is for invalid key presses. |
| 11:20.22 | Samot | i = invalid |
| 11:20.35 | jkroon | Samot, it'll work for that too depending on how you use it. |
| 11:21.05 | MajesticFudgie | apparently _. |
| 11:21.10 | MajesticFudgie | is whats needed |
| 11:21.25 | jkroon | MajesticFudgie, check the value of ${DIALSTATUS} post dial, or possibly ${HANGUPCAUSE}(=1) |
| 11:21.45 | MajesticFudgie | I've got it working now using _. lol |
| 11:21.46 | jkroon | ok, adious folks. i'm off. |
| 11:21.57 | ballock | jkroon: thanks |
| 11:22.31 | *** join/#asterisk scouture (~scouture@unaffiliated/scouture) |
| 11:22.52 | Lope | my caller calls, speaks to a member of the queue... now if my caller hangs up, my whole dialplan channel context seems to die instantly. |
| 11:23.00 | WIMPy | ballock: You just set your mobile to forward to your dongle if it's not reachable. That should do. |
| 11:23.01 | Lope | How can I make it keep processing stuff? |
| 11:23.07 | Samot | MajesticFudgie: When you say an invalid number you mean someone DIALED a bad number? |
| 11:23.10 | Lope | jkroon: cheers bro |
| 11:23.11 | MajesticFudgie | yup |
| 11:23.30 | Samot | That's going to be a HANGUPCAUSE |
| 11:23.43 | Samot | When you'll get when the Dial() result is returned. |
| 11:24.15 | Samot | s/When/What |
| 11:24.22 | ballock | WIMPy: well, not really. The problem is there is *some* range at home, so the phone still calls, but answering it requires my to walk to the window or outside, or the sound is... nearly there. |
| 11:25.29 | WIMPy | ballock: Then add forwarding on busy. That usually kicks in if you reject the call. |
| 11:25.39 | Lope | Can I keep my dialplan executing after the caller hangs up? |
| 11:25.44 | Samot | All _. is match any digit pattern and length. |
| 11:25.46 | WIMPy | Or your're lucky and your provider offers parallel forwarding. |
| 11:26.00 | Samot | Which means you let any type of number be dialed. |
| 11:26.22 | WIMPy | Not only digits/numbers. |
| 11:26.39 | Samot | That too. |
| 11:26.39 | WIMPy | That means you also catch the spechial extensions. |
| 11:26.40 | ballock | WIMPy: that sounds like a good idea. |
| 11:27.34 | Lope | Do I need a hangup handler if I want to process some stuff at the end of a call? |
| 11:27.40 | Lope | in my dialplan? |
| 11:28.10 | WIMPy | Depends on who hung up. |
| 11:29.07 | WIMPy | See Dial() for otions to send both call legs to the h extension or to continue in the dialplan. |
| 11:30.29 | *** join/#asterisk mub (~Jub@65-123-23-3.dia.static.qwest.net) |
| 11:30.42 | Lope | WIMPy: the caller hangs up? |
| 11:30.53 | Lope | WIMPy: I'm using a queue |
| 11:31.10 | Lope | the case I'm trying to deal with is if the caller hangs up while on a call in the queue. |
| 11:31.25 | Lope | I've got to do some post-processing of the channel. |
| 11:31.42 | Lope | is this syntax old? same => n,Set(CHANNEL(hangup_handler_push)=hdlr3,s,1(args)); |
| 11:31.55 | WIMPy | Never tried that, but sounds like a good use for a hangup handler. |
| 11:32.09 | Lope | Yeah it does. |
| 11:36.07 | *** join/#asterisk CeBe (~CeBe@85.214.9.200) |
| 11:36.16 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-lbrfyeeiyftrncic) |
| 11:36.57 | MajesticFudgie | Is there a sound clip that says invalid number rather than invalid extension? |
| 11:37.49 | *** join/#asterisk kolko (~kolko@46.48.58.17) |
| 11:38.45 | Lope | This hangup handler is not being executed. |
| 11:39.36 | Lope | I'm trying the first example on this page: https://wiki.asterisk.org/wiki/display/AST/Hangup+handlers any ideas? |
| 11:40.08 | elyob | Is there an obvious reason why my local network drops when my asterisk answers a call? i.e. all devices lose internet connections as soon as asterisk answers the call. Am running a test setup behind a BT Home Hub, and wonder if it's that at fault? |
| 11:40.40 | Lope | I hangup and nothing happens. the whole channel just dies. |
| 11:42.43 | Lope | When I run `core show hanguphandlers SIP/foo` it says there are no hangup handlers. So that would explain why they don't fire. What's a mystery is why it's not adding it. |
| 11:43.11 | Lope | oh, I need to load the app_stack module. |
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| 11:45.56 | Lope | awesome, it's working :) |
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| 11:49.11 | *** join/#asterisk degan6 (~degan6@c-73-151-71-161.hsd1.ca.comcast.net) |
| 11:54.40 | Lope | How can I get the context of the channel? |
| 11:56.12 | Lope | I don't see a channel var or function for it |
| 11:56.23 | Zogot | Lope: not ${CONTEXT}? |
| 11:56.56 | Lope | Will try. I didn't see it in ChanDump |
| 11:59.26 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:02.13 | Lope | I want to run my hangup handler when the caller hangs up (which is happening). But I also want to run it when they don't hangup, and I end the call by running Hangup. But in that case I get an error. ERROR[21071][C-00000003]: app_stack.c:594 gosub_exec: Attempt to reach a non-existent destination for Gosub: (Context:bcm_hangup_handler, Extension:s, Priority:1) |
| 12:02.55 | Lope | It seems like the Hangup app deletes the channel, and then the handler is left high and dry, trying to run a sub it can't find. |
| 12:03.54 | MajesticFudgie | elyob what BT Hub, I have a BT Hub 5 and it seems to be fine |
| 12:04.21 | Lope | Maybe I can run my hangup handler with the Gosub app. |
| 12:08.42 | Lope | Okay I got it working, nomatter who hangs up. |
| 12:11.21 | MajesticFudgie | Would this work with asterisk http://www.voipsupply.com/cisco-spa122 for incoming and outgoing to POTS |
| 12:15.20 | Lope | Can someone please help with Gosub? Description: Jumps to the label specified, saving the return address. Syntax: Gosub([[context,]exten,]priority[(arg1[,...][,argN])]) |
| 12:15.56 | Lope | At the moment I'm running Gosub(mycontext) but it's complaining that it can't find the LABEL mycontext in (the current context) |
| 12:16.39 | WIMPy | Where are extension and priority? |
| 12:17.13 | Lope | oh, I thought they were optional. |
| 12:17.24 | WIMPy | Where are the []? |
| 12:17.28 | Lope | I see now that the subroutine is like a virtual extension |
| 12:18.03 | WIMPy | Nor more virtual than anything else. It's a place in your dialplan. |
| 12:20.51 | Lope | hmm, I guess so. Thanks WIMPy. It's working now, no errors :) |
| 12:27.44 | Lope | Strange. Now when I run Hangup() it runs the hangup handler without any issues. No workarounds required. Maybe there was a mismatch in my name before. |
| 12:29.31 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:32.49 | [TK]D-Fender | MajesticFudgie, No. That lets you plug 2 phones in. No lines. |
| 12:33.04 | MajesticFudgie | ah |
| 12:33.17 | [TK]D-Fender | those are FXS ports on it. |
| 12:33.24 | MajesticFudgie | I need FXS and FXO? |
| 12:33.37 | [TK]D-Fender | No, so far just FXO |
| 12:33.42 | [TK]D-Fender | ~fxsfxo |
| 12:33.42 | infobot | [~fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 12:33.45 | MajesticFudgie | ah okie |
| 12:34.11 | MajesticFudgie | I'll need FXO then to send and recieve calls with the telco |
| 12:34.31 | Lope | The last thing I'd like to get is the duration that the caller spoke to a member of a queue for. Any ideas? |
| 12:34.32 | *** join/#asterisk Draecos (~Draecos@203-121-194-197.e-wire.net.au) |
| 12:35.01 | [TK]D-Fender | Lope, Read the log for it |
| 12:36.14 | Lope | TKD: perfect, thanks! |
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| 12:39.36 | Lope | Can I change the queue_log format so it only logs COMPLETEAGENT? the other lines are just a waste of disk space for me? |
| 12:40.33 | [TK]D-Fender | No. |
| 12:40.47 | [TK]D-Fender | If you're worried about disk space that much you have real problems |
| 12:41.02 | [TK]D-Fender | Feel free to prune the file yourself. |
| 12:41.33 | Lope | Okay. Can you suggest a safe method for truncating the log file without losing the most recent info? |
| 12:42.31 | [TK]D-Fender | How were you thinking of doing it that would do that? |
| 12:46.50 | Lope | I don't know. Normally I run truncate -s0 foo.log when I've processed the data in it. |
| 12:47.23 | Lope | But I can't script that to be run in my channel's post processing because other channels might be busy. |
| 12:48.07 | Lope | So I suppose I could run `truncate -s10K queue_log` |
| 12:48.36 | MajesticFudgie | Is it possible to forward to an extension like so. User calls a number starting with *, Asterisk forwards it to a specific set of extensions, stripping the * |
| 12:48.53 | [TK]D-Fender | no such thing as "forward". |
| 12:49.01 | MajesticFudgie | oh |
| 12:49.04 | [TK]D-Fender | you can Dial whatever you want, but don't call it forward. |
| 12:49.16 | MajesticFudgie | So I'd have to manually add extensions with the *? |
| 12:49.26 | [TK]D-Fender | You can also accept calls on whatever pattern you want. This has nothing to do with how you process the call |
| 12:49.27 | MajesticFudgie | one for 6001 and one for *6001 |
| 12:49.51 | [TK]D-Fender | You can't have one pattern that catches both of those that doesn't catch literally everything |
| 12:50.08 | [TK]D-Fender | Read your dialplan basics |
| 12:50.20 | MajesticFudgie | I'm setting up a seperate system for landline phones, dialing a number automatically calls outside and dialing with a * dials an extension |
| 12:50.23 | Lope | What I'm concerned about is a race condition: What if my truncate command starts, reads the file up to line x into a buffer and truncates the buffer. Another channel writes lines x+1 x+2 x+3. Then the truncate command writes it's (outdated) truncated buffer to the log file. Now I've lost the newest data? |
| 12:52.03 | [TK]D-Fender | MajesticFudgie, remember what I said about calling them "extensions". Don't. |
| 12:52.10 | MajesticFudgie | Well they are? |
| 12:52.10 | MajesticFudgie | lol |
| 12:52.16 | MajesticFudgie | Im editing extensions.conf |
| 12:52.19 | MajesticFudgie | What else would they be |
| 12:52.30 | [TK]D-Fender | extension = a line from extensions.conf |
| 12:52.33 | MajesticFudgie | Yup |
| 12:53.05 | [TK]D-Fender | * doesn't dial an "extension". YOU dial and extension. You keep trying to call a PHONE DEVICE an "extension". Stop. |
| 12:53.09 | MajesticFudgie | YES |
| 12:53.18 | MajesticFudgie | You're misunderstanding me |
| 12:53.27 | [TK]D-Fender | No, you're using the WRONG WORDS |
| 12:53.38 | [TK]D-Fender | Which we're trying to correct |
| 12:53.40 | MajesticFudgie | I'm trying to make an extension starting with * that strips the * and calls another set of extensions |
| 12:54.14 | [TK]D-Fender | You don't call going from one part of the dialplan to another "dialing" either |
| 12:54.37 | MajesticFudgie | You still know what I mean |
| 12:54.47 | [TK]D-Fender | Not even entirely |
| 12:55.11 | MajesticFudgie | I'm trying to explain what I wish to do |
| 12:55.16 | MajesticFudgie | I can't help I have no clue how to do it |
| 12:55.21 | [TK]D-Fender | your actual concept is pretty vague until we see what you do with it. And you've described this as though you intended 2 ways to do the same thing |
| 12:55.25 | MajesticFudgie | I'm trying to explain with what words I know |
| 12:55.54 | [TK]D-Fender | And I can tell you that practically no-one has their users dialing * in front to dial "direct" phone devices associated with their system. |
| 12:56.33 | [TK]D-Fender | the extensions most people use for local phones are usually 2-4 digits without any prefix |
| 12:57.04 | [TK]D-Fender | and since PSTN #'s are always longer there is no point in them. the local ones are already unique by length alone and you are jsut adding to what people have to dial for nothing |
| 12:57.18 | MajesticFudgie | I'll explain it as a setup. Two sets of devices. SIP Clients on Mobile Phones and Landline Phones. Two groups. The mobile group already have direct access to the extensions. E.g. 6002 for me. start a number with 9 for an outside line. Group 2, Landline phones, they automatically forward the number to an outside line just like a normal Landline phone using POTS, if they wanted to use extensions |
| 12:57.18 | MajesticFudgie | say to call 6002 they'd call *6002 |
| 12:57.24 | MajesticFudgie | Does what I'm wanting make any sense now? |
| 12:57.35 | MajesticFudgie | Regardless of whether it makes sense as a system |
| 12:59.28 | [TK]D-Fender | is that saying that you have a person we'll call "6002" that you want to have 2 different ways to reach them? |
| 12:59.41 | MajesticFudgie | yes |
| 12:59.53 | MajesticFudgie | depending if you use your sip client, as in your PC or mobile |
| 12:59.58 | MajesticFudgie | Or if you're using a house phone |
| 13:00.17 | MajesticFudgie | It's a home office setup, I dont want to distrupt my family too much |
| 13:00.45 | MajesticFudgie | But still provide an underlying way to reach any phone if required |
| 13:01.20 | MajesticFudgie | Apologies for being a pain btw, just trying to make sense of everything still |
| 13:02.04 | MajesticFudgie | http://paste.ubuntu.com/16201232/ - Is my extensions.conf it shows what I'm trying to do. Although I'm sure theres 500 ways to do it better |
| 13:02.16 | [TK]D-Fender | So your users are going to do what? Dial 6002 and when there is no answer hangup and call *6002 to keep trying to find you? |
| 13:02.54 | MajesticFudgie | No, it depends on the phone |
| 13:03.06 | [TK]D-Fender | What do you mean depends on the phone? |
| 13:03.15 | MajesticFudgie | Depends on its purpose |
| 13:03.18 | [TK]D-Fender | You just said you're making 2 ways to reach each person |
| 13:03.22 | MajesticFudgie | yes |
| 13:03.28 | MajesticFudgie | If its an office phone you dial 6002 |
| 13:03.36 | [TK]D-Fender | 1 via dialing a SIP device associated to them, and another that will call their PSTTN home phone #'s |
| 13:03.37 | MajesticFudgie | if Its just a house phone the family uses they dial *6002 |
| 13:03.48 | MajesticFudgie | yeah, I think so |
| 13:03.55 | [TK]D-Fender | you just told me that people will have both |
| 13:04.04 | [TK]D-Fender | so not "just" home phone |
| 13:04.08 | [TK]D-Fender | You aren't being clear |
| 13:04.09 | MajesticFudgie | they will if they have their mobile |
| 13:04.29 | MajesticFudgie | It's an attempt to make the house phone act like a house phone |
| 13:04.37 | MajesticFudgie | But still have access to the extensions |
| 13:05.00 | MajesticFudgie | I could just keep it connected to the telcom but I'd then be unable to access any of the pbx system |
| 13:05.02 | [TK]D-Fender | ? |
| 13:05.09 | [TK]D-Fender | how are you affecting a land-line phone at all? |
| 13:05.20 | [TK]D-Fender | their telco determines their calling and your PBX has nothing to do with this |
| 13:05.35 | MajesticFudgie | If I didn't make a seperate group everyone would need to dial 9[insert number for outgoing to telcom] |
| 13:05.41 | MajesticFudgie | and would annoy the family |
| 13:06.09 | [TK]D-Fender | <[TK]D-Fender> ? |
| 13:06.09 | [TK]D-Fender | <[TK]D-Fender> how are you affecting a land-line phone at all? |
| 13:06.09 | [TK]D-Fender | <[TK]D-Fender> their telco determines their calling and your PBX has nothing to do with this |
| 13:06.10 | [TK]D-Fender | ^^^^^^^ |
| 13:06.17 | MajesticFudgie | yes |
| 13:06.23 | MajesticFudgie | But I want it to have PBX access |
| 13:06.27 | [TK]D-Fender | makes NO sense |
| 13:06.29 | MajesticFudgie | without hindering its original use |
| 13:06.35 | [TK]D-Fender | You can't control some phone on the PSTN |
| 13:06.54 | MajesticFudgie | Idk how else to explain it |
| 13:07.19 | [TK]D-Fender | If I have a phone plugged into a dumb phone line provided by my phone company then it has NOTHING to do with my PBX. |
| 13:07.24 | MajesticFudgie | I know |
| 13:07.32 | [TK]D-Fender | So how are you taking magical control over it? |
| 13:07.34 | MajesticFudgie | Thats the idea, it needs to act like a dumb phone in the PBX |
| 13:07.35 | [TK]D-Fender | it;'s not ON your PBX |
| 13:07.42 | [TK]D-Fender | what "it"? |
| 13:07.44 | MajesticFudgie | but I can still ring it fro mthe pbx |
| 13:07.49 | MajesticFudgie | The landline phone |
| 13:07.50 | [TK]D-Fender | you said some guy's HOME PHONE |
| 13:07.54 | MajesticFudgie | Yes |
| 13:07.58 | MajesticFudgie | ignore me |
| 13:08.01 | MajesticFudgie | I'll reword it |
| 13:08.04 | [TK]D-Fender | that's his TELCO's job |
| 13:08.54 | MajesticFudgie | The idea is, my room has an office phone. allows internal calling directly. The family's home phone in the passage way needs to act like a normal phone that contacts the telco |
| 13:09.06 | MajesticFudgie | But if they want to ring up to my room they'd need to dial the extension for me |
| 13:09.26 | MajesticFudgie | However at the current state, the _. extension would try to forward all calls through the FXO/FXS Device |
| 13:09.32 | [TK]D-Fender | You are talking about 2 completely different things |
| 13:09.38 | [TK]D-Fender | and screwing them up bad |
| 13:09.46 | MajesticFudgie | I'd need a way to dial the extensions |
| 13:09.57 | MajesticFudgie | without it being caught by _. |
| 13:10.01 | [TK]D-Fender | "The family's home phone" this is NOT A LINE |
| 13:10.21 | MajesticFudgie | I never said its a line |
| 13:10.24 | MajesticFudgie | I said its a phone |
| 13:10.34 | MajesticFudgie | It'll be a device connect but treated differently |
| 13:10.45 | [TK]D-Fender | you keeps saying landline in relation to it |
| 13:10.52 | MajesticFudgie | Its what we call it in the UK |
| 13:10.54 | MajesticFudgie | Landline phone |
| 13:10.57 | MajesticFudgie | family phone |
| 13:10.59 | [TK]D-Fender | And how is THAT actually connected to your PBX? |
| 13:11.02 | MajesticFudgie | aka not my phone in my room |
| 13:11.14 | MajesticFudgie | It'd be connected like a normal VOIP Phone |
| 13:11.23 | [TK]D-Fender | MajesticFudgie> Landline phone <- no such thing. the line is the lin, the phone is the phone. |
| 13:11.25 | MajesticFudgie | I jusst have to make it act like its not unless I want it to |
| 13:11.45 | [TK]D-Fender | <MajesticFudgie> It'd be connected like a normal VOIP Phone <- how? |
| 13:11.51 | MajesticFudgie | Because it would be a VOIP Phone |
| 13:12.05 | [TK]D-Fender | you just said it was a LANDLINE phone |
| 13:12.08 | [TK]D-Fender | ^^^ |
| 13:12.08 | MajesticFudgie | Yes |
| 13:12.12 | MajesticFudgie | Thats what you call it here |
| 13:12.18 | MajesticFudgie | Thats what I'd know it as |
| 13:12.19 | [TK]D-Fender | <MajesticFudgie> Because it would be a VOIP Phone <- then it ISN'T |
| 13:12.25 | MajesticFudgie | regardless of it being connected to the Telco |
| 13:12.25 | [TK]D-Fender | How the hell is it BOTH |
| 13:12.40 | MajesticFudgie | ._. |
| 13:12.52 | MajesticFudgie | Terminology, I don't mean literally a landline phone |
| 13:13.03 | MajesticFudgie | I mean the phone thats percieved as having the purpose of calling outbound to the telco |
| 13:13.13 | [TK]D-Fender | stop talking about it generically |
| 13:13.14 | MajesticFudgie | I'd casually refer to it as a landline phone |
| 13:13.39 | [TK]D-Fender | you are making things WORSE. Why kion do fstupid hardware are you SPECIFICALLY using and what is it specifically PLUGGED INTO? |
| 13:13.50 | MajesticFudgie | I'll make a diagram |
| 13:25.58 | *** join/#asterisk obelixBE (~obelix@d5152CCD8.static.telenet.be) |
| 13:26.09 | MajesticFudgie | https://surl.im/i/pnbkj.png This make make some sense, the lines aren't supposed to represent an actual physical connection |
| 13:26.49 | MajesticFudgie | Phones starting with 6 could call the phone starting with 5 just by typing 5001, an example for Phone A is calling *5001 could call its self |
| 13:26.54 | MajesticFudgie | which is ofc pointless |
| 13:27.21 | MajesticFudgie | and calling 9123 would call 123 on the telco on Phone B and C |
| 13:27.31 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:27.32 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:27.32 | [TK]D-Fender | A B & C are all behind that Asterisk box |
| 13:27.35 | MajesticFudgie | yup |
| 13:27.39 | [TK]D-Fender | why the hell do you need 2 ways to call them? |
| 13:27.42 | MajesticFudgie | But A is treated a little different |
| 13:27.51 | [TK]D-Fender | WHY? |
| 13:27.52 | MajesticFudgie | Because A isn't to be treated like an office phone |
| 13:28.02 | [TK]D-Fender | means nothing |
| 13:28.11 | [TK]D-Fender | If you want to dial out then jsut a normal number |
| 13:28.13 | MajesticFudgie | B and C are office phones, they'd be in 'office rooms' |
| 13:28.21 | [TK]D-Fender | why wuould you need 2 ways to call A from B? |
| 13:28.28 | [TK]D-Fender | Why would you need 2 ways to call B from A? |
| 13:28.38 | MajesticFudgie | You cant |
| 13:28.45 | MajesticFudgie | You call B from A via *6001 |
| 13:28.55 | [TK]D-Fender | I fail to see why need 2 ways for any of these cases |
| 13:29.02 | [TK]D-Fender | Why not just dial 6001? |
| 13:29.07 | [TK]D-Fender | Why the dumb * in front? |
| 13:29.11 | MajesticFudgie | Because it'd be forwarded to the telco |
| 13:29.14 | [TK]D-Fender | what's the POINT? |
| 13:29.19 | MajesticFudgie | The idea is my mum doesnt have to press 9 on Phone A |
| 13:29.28 | MajesticFudgie | But on Phone B in my room I do press 9 |
| 13:29.36 | MajesticFudgie | To be able to call the TELCO |
| 13:29.37 | [TK]D-Fender | You never need a stupid 9 in the first place |
| 13:30.00 | [TK]D-Fender | And you can dial whatever you want and do whatever you want for ANY device |
| 13:30.01 | MajesticFudgie | I'm just used to it in offices |
| 13:30.12 | [TK]D-Fender | There is still no need for any of that |
| 13:30.29 | MajesticFudgie | How would I do it anyway |
| 13:30.54 | [TK]D-Fender | PATTERNS. 6001 is not a number on the PSTN |
| 13:31.02 | MajesticFudgie | Yes thats what I want |
| 13:31.05 | [TK]D-Fender | so maky your extension tto match it and just DO IT |
| 13:31.13 | MajesticFudgie | take include => blah |
| 13:31.19 | MajesticFudgie | I'd want anything in [blah] |
| 13:31.24 | MajesticFudgie | But with a number in front |
| 13:31.33 | [TK]D-Fender | No need for anything in front |
| 13:31.35 | *** join/#asterisk theron (~theron@2620:10d:c091:200::2:12ab) |
| 13:31.39 | MajesticFudgie | What if I wanted to? |
| 13:31.42 | [TK]D-Fender | You are adding crap for no good reason |
| 13:31.53 | [TK]D-Fender | and either way we are talking about what you'd dial to dial BETWEEN your local phones |
| 13:32.00 | [TK]D-Fender | which doesn't matter to that anyway |
| 13:32.07 | [TK]D-Fender | Youp want to dial 6001 to call A... then DO IT |
| 13:32.20 | [TK]D-Fender | Youp want to dial 6001 to call BA... then DO IT (corrected) |
| 13:32.22 | [TK]D-Fender | B |
| 13:32.23 | [TK]D-Fender | gah |
| 13:32.29 | MajesticFudgie | exten => _X.,n,Goto(from-internal,${NUMTOCALL},1) ; |
| 13:32.33 | MajesticFudgie | apparently is what I'd need |
| 13:32.33 | [TK]D-Fender | no need for another infrom of those 4-digit numbers |
| 13:32.44 | [TK]D-Fender | taht is a STUPID pattern |
| 13:32.55 | [TK]D-Fender | Since when is 22 a valid # to call on the PSTN? |
| 13:33.28 | [TK]D-Fender | You need to make actual patterns that make actual sense |
| 13:34.40 | [TK]D-Fender | And regardless you can put extensions used to dial local devices in 1 context, and extensions used to dial out to the PST in another and INCLUDE those 2 contexts in a 3rd context in that order so those local #'s don't get routed out |
| 13:35.08 | [TK]D-Fender | Telco #'s look like real #'s which look nothing like the one's you are associating with A,B, and C |
| 13:36.42 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 13:37.16 | [TK]D-Fender | Prefixes are a complete waste of time. |
| 13:37.42 | MajesticFudgie | Well I'm just a hobbyist trying to setup Asterisk, Idk any proper practices or anything. I'm just trying my best |
| 13:38.55 | [TK]D-Fender | you have 3 extension to dial your local phones and at least 1 real decent pattern to route thing that should go to the PSTN out via whatever you use for that |
| 13:39.08 | saint_ | hi all - I have my Asterisk connected via SIP to another PBX. If my asterisk call an extension on the other PBX, and the calle puts the call on hold, in on the asterisk user side I hear shit music. It's a mix of both asterisk MOH and remote PBX moh . is this a bug, or is there a way to fix this ? |
| 13:39.09 | [TK]D-Fender | You complicated this all for nothing |
| 13:39.22 | *** join/#asterisk fvox13 (~stevensmi@cpe-72-225-42-19.rochester.res.rr.com) |
| 13:39.44 | MajesticFudgie | mhm |
| 13:39.53 | [TK]D-Fender | And you called A a "landline phone". You never use the word "land-line" when you aren't catually calling out to the public phone network to a telco-host phone |
| 13:40.14 | [TK]D-Fender | land-line = line over land to TELCO |
| 13:40.56 | MajesticFudgie | Yeah, I'm just used to my countries slang names |
| 13:41.14 | [TK]D-Fender | a phone is not a line. |
| 13:41.16 | MajesticFudgie | Even if the phone was sat in the middle of the room connected to nothing we'd still call it a landline phone lol |
| 13:41.27 | [TK]D-Fender | So choving the word "line" in there is a serious mistake |
| 13:41.47 | MajesticFudgie | Blame the UK ok |
| 13:41.48 | [TK]D-Fender | You don't use the word "land line" unless it's plugged into a TELCO LINE |
| 13:42.06 | [TK]D-Fender | If I have a phone sitting unplugged in the middle of the room... it's just a DUMB PHONE. |
| 13:42.15 | MajesticFudgie | I'm fully away |
| 13:42.17 | MajesticFudgie | aware* |
| 13:42.34 | [TK]D-Fender | If that phone is plugged into your PBX then there is no stupid land line. |
| 13:42.45 | [TK]D-Fender | So how is that thing going to talk to your PBX? |
| 13:43.07 | *** join/#asterisk FitoVbonito (~adolfo@nat2g174-127.wi-fi.upv.es) |
| 13:43.16 | MajesticFudgie | You're taking the way I speak too literally, thats all |
| 13:43.32 | [TK]D-Fender | Words have meaning |
| 13:43.40 | MajesticFudgie | I'm not trying to be an idiot that doesnt know anything. It's just how I communicate what I'm trying to mean |
| 13:44.42 | [TK]D-Fender | You'll need to fix your terms. The picture helped |
| 13:45.24 | MajesticFudgie | I know, sorry about that and good. I didn't know how else to explain it lol |
| 13:45.29 | MajesticFudgie | I'm terrible with words tbh |
| 13:45.40 | MajesticFudgie | I'm best at physically showing someone what I mean |
| 13:46.14 | MajesticFudgie | Anyway i need to get going somewhere, Thanks for helping out. apologies for stressing you. ttyl |
| 13:50.15 | [TK]D-Fender | we'll pick up on this later and help get the plan straightened out |
| 13:52.28 | *** join/#asterisk KerioMorgan (~Adium@gw-us.kerio.com) |
| 13:52.31 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 13:56.02 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 14:00.08 | *** join/#asterisk theron (~theron@2620:10d:c091:200::d:365) |
| 14:00.37 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 14:00.49 | *** join/#asterisk LooserOuting (~LooserOut@p578aa533.dip0.t-ipconnect.de) |
| 14:05.51 | LooserOuting | does somebody know how long a to-tag can be? i think i have a situation here where asterisk can't find a transaction because the to-tag is too long |
| 14:07.41 | LooserOuting | i am using chan_sip and I receive those long to-tags from my provider |
| 14:12.29 | scv | well |
| 14:12.37 | scv | thats neat |
| 14:13.33 | scv | under 13.8.x if you have parkcall in [featuremap] in features.conf (i.e. old legacy config that shouldn't be present) |
| 14:14.04 | scv | you get silent errors when bridging channels and the call just drops |
| 14:14.06 | scv | without any error |
| 14:14.37 | scv | under debugging there is a notice about that deprecated option but it isn't clear at all that's why the endpoint fails to join the bridge |
| 14:20.48 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 14:25.34 | *** join/#asterisk kharwell (kharwell@nat/digium/x-cscnumbevykeekru) |
| 14:32.13 | cresl1n | waves to kharwell |
| 14:32.16 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 14:32.32 | *** join/#asterisk iamb (~iamb@unaffiliated/iamb) |
| 14:32.47 | kharwell | cresl1n: hiya |
| 14:33.52 | file | wobbles |
| 14:34.56 | Kobaz | what do you guys do for asterisk on multiple internets |
| 14:35.43 | Kobaz | i'm thinking of writing a script that'll detect if one is down and then it'll change the default gateway, and then set externip in sip as well |
| 14:36.40 | Kobaz | or, i can run two asterisk's in virtual machines each with one of the gateways, and then dial through one, then the other |
| 14:36.59 | Kobaz | that would probably be the least trouble free to just have both always up |
| 14:37.13 | Kobaz | and then reinvite sip so it's not double processing the media |
| 14:38.01 | saint_ | anyone has an idea why when asterisk call another pbx , and the other pbx puts the call on hold, the caller on asterisk side hears both local and remote MOH at the same time ? |
| 14:44.40 | *** join/#asterisk theron (~theron@199.201.64.132) |
| 15:13.11 | *** join/#asterisk drale2k (~drale2k@46.101.230.32) |
| 15:20.16 | *** join/#asterisk hirogen (c22afc44@gateway/web/freenode/ip.194.42.252.68) |
| 15:20.18 | hirogen | hi |
| 15:21.30 | hirogen | you know in avaya site administration 6.0 ? |
| 15:21.52 | hirogen | can you run reports to see if tehre are any available licences or rather to see when an extension number was last used so any old ones we could reuse ideally? |
| 15:35.06 | *** join/#asterisk jriffel73 (ae4fbb4c@gateway/web/freenode/ip.174.79.187.76) |
| 15:35.31 | *** join/#asterisk cip (ad413e2f@gateway/web/freenode/ip.173.65.62.47) |
| 15:35.47 | jriffel73 | Howdy CIP. |
| 15:36.03 | cip | HOwdy |
| 15:40.37 | cip | Is there perhaps someone who might guide me through troubleshooting a basic problem was Asterisk 1.8.11 ? The most basic commands don't seem to be available such as "core show version" |
| 15:41.08 | jriffel73 | Is this on an existing server that ever worked, or a new installatino? |
| 15:41.21 | cip | existing, was working before the weekend. |
| 15:41.37 | jriffel73 | So its not a new install? |
| 15:41.50 | cip | No sir |
| 15:42.08 | jriffel73 | Did you verify the processes are running? ps auxww | grep asterisk ? |
| 15:42.44 | cip | asterisk is running, yes. I can asterisk -r and type commands; it's just that many commands aren't available |
| 15:42.47 | jriffel73 | Also, you might try checking to see the ports are bound properly - netstat -an | more |
| 15:43.12 | jriffel73 | Look for ports bound to 5060 for SIP and I think 5036 for an asterisk management port |
| 15:43.37 | cip | 5038 is bound to 0.0.0.0, 5060 is not bound as the sip module isn't loaded. |
| 15:43.47 | jriffel73 | So you can get into asterisk -r ... |
| 15:43.52 | cip | yes |
| 15:44.14 | jriffel73 | grep -i error /var/log/asterisk/full |
| 15:44.17 | jriffel73 | anything good in there? |
| 15:44.24 | cip | Nope |
| 15:44.52 | jriffel73 | Hrm... |
| 15:45.37 | cip | Asterisk process is running at around 50% cpu all the time yet no modules are loaded |
| 15:46.34 | jriffel73 | I dunno sir... that is getting to the end of my skillz. |
| 15:47.11 | cip | Thanks, anyway. Anyone else ? |
| 15:51.12 | acidfu_ | cip you should attaching a profiler |
| 15:51.17 | acidfu_ | to see what functions are being called |
| 15:51.40 | acidfu_ | and what is your CPU ? and how do you "see" that it is taking 50% of your cpu |
| 15:51.54 | cip | via top |
| 15:51.57 | acidfu_ | 50% can look much, but if nothing else is running... |
| 15:52.07 | acidfu_ | 50% relative to almost nothing is not much |
| 15:52.31 | cip | Sure, understood though it never used to be that way |
| 15:52.37 | [TK]D-Fender | Stop * and restart it\ |
| 15:52.45 | cip | Done that multiple times today |
| 15:52.51 | [TK]D-Fender | prove that it's starting up properly |
| 15:52.54 | [TK]D-Fender | Showe us |
| 15:53.16 | [TK]D-Fender | PASTEBIN the shutdown, and then starting it MANUALLY (not as a service) |
| 15:53.19 | [TK]D-Fender | ~pb |
| 15:53.20 | 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. |
| 15:53.20 | [TK]D-Fender | ^^^ |
| 15:53.28 | cip | Interestingly, "core stop now" doesn't work. |
| 15:53.56 | [TK]D-Fender | So far that doesn't sound like it agrees with "Done that multiple times today" |
| 15:54.00 | cip | ok, you want to see the log from time of shutdown ? |
| 15:54.08 | [TK]D-Fender | I want to see you do it NOW |
| 15:54.33 | syadnom | I'm trying to figure out how to get the jitter buffer in ast 13.7 to log. I have it enabled in sip.conf and asterisk console sip show settings has: Jitterbuffer log: Yes |
| 15:54.33 | syadnom | <PROTECTED> |
| 15:54.40 | syadnom | nothing in /tmp for jitter buffer logs though.. |
| 15:55.23 | *** join/#asterisk BarthezZ (~bart@monitoring.deheij-ict.nl) |
| 15:56.05 | cip | D-Fender, I can obviously shut it down with /etc/init.d/asterisk stop, but "core stop now" results in "No such command 'core stop now' (type 'core show help core' for other possible commands)" |
| 15:56.27 | [TK]D-Fender | You may have issued a command but that doesn't prove it happened. |
| 15:56.43 | [TK]D-Fender | We need actual proof |
| 15:56.57 | [TK]D-Fender | stop it. Prove that it has completely stopped and no processes running. |
| 15:57.07 | cip | Standby, I'll use paste a log to one of those services. |
| 15:57.11 | [TK]D-Fender | Start it manually (not as a daemon) and show us all of this. |
| 15:57.19 | [TK]D-Fender | logs = trash |
| 15:57.22 | [TK]D-Fender | live CLI only |
| 15:57.26 | cip | ok |
| 16:02.29 | scv | [TK]D-Fender you really are rather pleasant, in my two days of idling this channel you've successfully been a twat about 5 dozen times |
| 16:02.41 | scv | classic f/oss attitude :) |
| 16:02.44 | *** part/#asterisk scv (solar@162.217.176.109) |
| 16:02.51 | [TK]D-Fender | :) |
| 16:10.49 | jriffel73 | Well what do you expect for free? |
| 16:10.58 | jriffel73 | I'll take good answers from a twat any day |
| 16:12.02 | cip | http://pastebin.com/Pqep0HNm |
| 16:16.12 | [TK]D-Fender | a few things. You skipped the part of showing your actually shupping down * and proving it was down first and we also don't see how you started that, all of which I asked for. |
| 16:16.48 | [TK]D-Fender | But since you're there now.... you still haven't shown if the commands that didn'tt work previously were now working... |
| 16:16.58 | [TK]D-Fender | So .... are they? |
| 16:17.33 | cip | No they are not |
| 16:18.14 | cip | ip-172-30-3-123*CLI> core show version No such command 'core show version' (type 'core show help core' for other possible commands) |
| 16:18.15 | [TK]D-Fender | Lets repeat the process and show the complete backup as requested along with the attempts following CLI finishing. |
| 16:18.48 | cip | ok standby, I'll get you the shutdown |
| 16:18.55 | cip | and a proces list if you'd like |
| 16:19.03 | [TK]D-Fender | that's what I asked for twice now. |
| 16:20.58 | cip | Actually nothing is produced in the logfile when shutting down via /etc/init.d/asterisk stop. And since I cannot execute a shutdown via 'core shutdown now', there is no other way to stop it. |
| 16:21.46 | [TK]D-Fender | kill <--------------------- |
| 16:21.50 | [TK]D-Fender | kill begs to differ |
| 16:22.05 | [TK]D-Fender | that's why I asked for proof that it is dead. |
| 16:22.45 | cip | ps ax | grep asterisk reveals no processes. Do you want me to paste the output of ps ax ? |
| 16:24.11 | [TK]D-Fender | ps -A |
| 16:24.23 | [TK]D-Fender | scan it globally properly |
| 16:24.36 | [TK]D-Fender | cast a wider net and make sure other funny bits aren't going wrong. |
| 16:24.52 | [TK]D-Fender | then start it manually and show the full command. All of it. not tiny bits |
| 16:25.58 | cip | http://pastebin.com/KkTP8xJ8 |
| 16:27.22 | [TK]D-Fender | Ok, let me be clear... I want to see the COMMAND in there as well. |
| 16:27.24 | [TK]D-Fender | not justt ooutput |
| 16:28.37 | [TK]D-Fender | Show the manual startup. Not as a daemon.... |
| 16:35.45 | cip | http://pastebin.com/jN3LDGJ0 |
| 16:42.57 | [TK]D-Fender | [1;30m == [0mParsing '/etc/asterisk/modules.conf': [1;30m == [0mFound |
| 16:42.58 | [TK]D-Fender | [2016-05-03 12:31:52] [1;33mNOTICE[0m[18548]: [1;37mloader.c[0m:[1;37m1129[0m [1;37mload_modules[0m: 2 modules will be loaded. |
| 16:43.06 | [TK]D-Fender | 2 modules? that does NOT look sane |
| 16:43.32 | [TK]D-Fender | and you STILL didn't show the command attempts at the end of that |
| 16:44.27 | [TK]D-Fender | pastebin your modules.conf, every file it INCLUDES, "ls -la /etc/asterisk", and the output of "core show modules" |
| 16:45.21 | cip | ok, will do... Standby |
| 16:53.38 | cip | http://pastebin.com/4uhqc3bu |
| 16:57.04 | [TK]D-Fender | Something is pretty screwed up there... |
| 16:57.24 | [TK]D-Fender | "ls -la /usr/lib/asterisk/modules" |
| 16:59.31 | cip | http://pastebin.com/KJPTVzji |
| 17:01.52 | *** join/#asterisk raspberrypifan (~raspberry@2604:2000:6016:be00:3db0:9205:24f6:633f) |
| 17:03.36 | [TK]D-Fender | hrm |
| 17:04.05 | cip | One thing that we did observe on the system is that a file named ast-ami-XXXXX was created in the /tmp folder that was almost 4GB and it had run teh server out of disk space. The file was owned and created by the asterisk process. I'm wondering if the config files were corrupted when the disk became full. |
| 17:04.15 | cip | I did delete that file |
| 17:04.58 | [TK]D-Fender | have you completely rebooted the machine and verified free disk space? |
| 17:05.35 | cip | I have not rebooted the machine since, but df -h shows 4.3G available |
| 17:06.10 | [TK]D-Fender | that;'s a start. reboot the whoel thing then... |
| 17:06.15 | [TK]D-Fender | let's see if something else clears |
| 17:06.25 | [TK]D-Fender | just before you do though... |
| 17:06.37 | [TK]D-Fender | Since you seem to be running a FreePBX install : amportal chown |
| 17:06.38 | [TK]D-Fender | first |
| 17:06.40 | [TK]D-Fender | then restart |
| 17:06.51 | [TK]D-Fender | to give it the best chance of a clean sttartup |
| 17:07.13 | cip | ok |
| 17:07.37 | [TK]D-Fender | do that then on reboot go check your CLI immediately |
| 17:09.20 | raspberrypifan | im gonna chown you so hard |
| 17:09.29 | jriffel73 | right in the pi hole |
| 17:09.41 | raspberrypifan | lol |
| 17:19.42 | cip | Same result. |
| 17:20.25 | [TK]D-Fender | ok, I'm at a bit of a loss... |
| 17:20.55 | cip | ok.... Think I'm going to re-install the machine, assuming I can get my freepbx config to download |
| 17:21.44 | cip | Thanks for your help anyway. |
| 17:22.31 | mub | cip: You can also try to get help from #freepbx if you haven't already |
| 17:22.37 | mub | If this is a freepbx machine |
| 17:23.10 | jriffel73 | I bet cip wears a flesh colored unitard |
| 17:24.02 | *** join/#asterisk jkroon (~jkroon@uls-154-73-32-14.wall.uls.co.za) |
| 17:24.21 | cip | You bet! |
| 17:25.29 | [TK]D-Fender | I'd recommend against reinstalling that though |
| 17:25.32 | [TK]D-Fender | 1.8 is LONG dead |
| 17:28.15 | cip | If I do, it'll be the latest and greatest. |
| 17:29.02 | file | 1.8 has been gone for... geez, 6+ months |
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| 17:37.31 | mub | 13 is where it's at |
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| 17:38.19 | mub | I don't really understand why the versions jump from 1.X to 10.X |
| 17:38.52 | mub | Like, in 2010 there was 1.8.x, then in 2011 there was 10.x |
| 17:46.54 | file | the versioning scheme changed |
| 17:46.58 | file | it is unlikely to ever change again |
| 17:47.04 | [TK]D-Fender | because 1.1000 will start looking stupid |
| 17:47.23 | [TK]D-Fender | where in normal math 1.10 = 1.1 and we already had a 1.4 |
| 17:47.45 | *** join/#asterisk Kevin` (~kevin@99-96-57-112.lightspeed.milwwi.sbcglobal.net) |
| 17:47.50 | [TK]D-Fender | so it'd look silly to reduce to a number that looks lower |
| 17:48.53 | Kevin` | how can I enable or disable debug logging for a specific module/file? |
| 17:49.23 | Kevin` | I would espect something like 'core set debug 9 http.c' but that doesn't work |
| 17:50.13 | [TK]D-Fender | you can't |
| 17:52.17 | Kevin` | what does that option do per-file then? |
| 17:53.59 | [TK]D-Fender | Where do we see it doing anything "per file"? |
| 17:54.49 | Kevin` | somegen-dev*CLI> core set debug 1 |
| 17:54.49 | Kevin` | Core debug was OFF and is now 1. |
| 17:54.49 | Kevin` | somegen-dev*CLI> core set debug 2 http.c |
| 17:54.49 | Kevin` | Core debug was 9 and has been set to 2 for 'http.c'. |
| 17:55.36 | [TK]D-Fender | it certainly shouldn't be referencing a ".c" file |
| 17:55.41 | [TK]D-Fender | because * can't load ".c" |
| 17:55.43 | [TK]D-Fender | perhaps .so |
| 17:55.55 | [TK]D-Fender | Which is still new to me.... |
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| 18:19.19 | *** join/#asterisk jms77 (17f64122@gateway/web/cgi-irc/kiwiirc.com/ip.23.246.65.34) |
| 18:19.41 | jms77 | I have a question about configuring trunks to use tcp |
| 18:19.49 | jms77 | I have a trunk with the transport specified to a tcp transport |
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| 18:20.13 | jms77 | In the aor I have the contact set to 'sip:<ip-address>:5070' |
| 18:20.32 | jms77 | when I attempt to make a call I get the error jsip: tsx0x7f2644003 .Failed to send Request msg OPTIONS/cseq=62700 (tdta0x7f26440012f0)! err=171060 (Unsupported transport (PJSIP_EUNSUPTRANSPORT)) |
| 18:20.35 | file | you need to have \;transport=tcp |
| 18:20.40 | file | at the end of the URI |
| 18:20.43 | jms77 | If I do that |
| 18:20.54 | jms77 | I get a failed to parse contact |
| 18:21.08 | jms77 | [May 3 18:17:23] ERROR[14726]: res_pjsip/location.c:397 permanent_uri_handler: Permanent URI on aor 'oracle' with contact 'sip:199.233.226.216:5070\' failed to parse |
| 18:21.24 | file | how are you configuring it? in a database? |
| 18:21.27 | jms77 | yeagh |
| 18:21.31 | file | then remove the \ |
| 18:21.37 | jms77 | then i get |
| 18:21.40 | file | the \ is for .conf files so the ";" is not treated as starting a comment |
| 18:21.52 | jms77 | contact 'transport=tcp' failed to parse |
| 18:22.49 | file | that sounds as though it is treating it as if there are multiple contact fields... separated by ; |
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| 18:23.57 | jms77 | hmm |
| 18:28.28 | mub | .p g 2 |
| 18:28.30 | mub | woops |
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| 18:39.56 | jms77 | So I just switched it from a database driven config to a conf file |
| 18:40.02 | jms77 | but that's only a temporary solution |
| 18:40.30 | jms77 | What do I do, submit a bug report, email the developers or something? |
| 18:40.48 | file | I'd suggest identifying whether the issue is with Asterisk itself, or if it's the database first |
| 18:41.00 | jms77 | k |
| 18:41.14 | file | if it's Asterisk then you can file an issue on https://issues.asterisk.org/jira with complete details |
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| 19:46.06 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.8.2 (2016/04/20), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 20:48.38 | jge | X-Rob: not sure what did it but memory was pegged 100 percent, load looked high too so I uped ram to 1GB and now it's happy |
| 20:48.49 | jge | that explains the reboot earlier helping too.. |
| 20:49.01 | jge | it was running on 1vCPU/512 |
| 21:12.22 | X-Rob | wonders why he was pinged |
| 21:12.47 | X-Rob | jge: I don't know why you told me that, but thank you. |
| 21:13.38 | X-Rob | At least I don't have an IRC nick that is a commonly used IT term, like drive, or net, or file. What sort of crazy person would pick that as an IRC nick? |
| 21:21.46 | cresl1n | file is a personal favorite of mine |
| 21:22.14 | file | what's fun is on my laptop the Windows build is a bit wonky with sound - so if you mention my name at the same time as I get an email the audio combines and forms a horrible sound |
| 21:22.51 | cresl1n | wonders how he can arrange that to happen - reliably |
| 21:23.05 | file | cresl1n, you can't |
| 21:23.16 | file | my Digium email syncs at a rate of 1 minute, it's not instant :P |
| 21:23.22 | cresl1n | Ooooh |
| 21:23.27 | cresl1n | that makes it even easier |
| 21:23.53 | cresl1n | :-P |
| 21:24.06 | file | schedules more meetings for cresl1n |
| 21:24.18 | cresl1n | You can't make it any worse than it already is :-) |
| 21:24.37 | file | sounds like someone wants to have meetings with people in Europe and Asia |
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| 21:30.29 | V5_12R\X | Whats the best way to make a suggestion for additions to the make install script? I think that ldconfig should be added to the list of tasks |
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| 21:34.36 | WIMPy | Why do you (think you) need ldconfig? |
| 21:40.00 | Silicon | has anyone got pjsip asterisk-devicestate |
| 21:40.04 | Silicon | working |
| 21:40.13 | Silicon | i'm having a tough time trying to figure it out : |
| 21:40.35 | Silicon | https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP |
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| 22:15.36 | MajesticFudgie | I'm here to annoy you all again, can someone point me in the right direction to link Skype's SIP as a device/peer |
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| 22:29.51 | V5_12R\X | WIMPy for debian installs if you dont do an ldconfig you will get an error message. I'll have to re-install here to replicate it, but asterisk claims it's missing libasteriskssl.so |
| 22:30.05 | V5_12R\X | Silicon did you ever get that working> |
| 22:30.58 | MajesticFudgie | Anyone in here used SipGate before? |
| 22:32.48 | WIMPy | MajesticFudgie: What's the real question? |
| 22:33.13 | MajesticFudgie | Is it any good and does it work well with Asterisk :P |
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| 22:33.56 | WIMPy | It's an ITSP like any other and as teh name sugests, they use SIP, which Asterisk can do as well. |
| 22:35.14 | MajesticFudgie | okie |
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