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| 00:46.39 | lfamorim | Hello! Someone know a good SIP/VoIP provider who supports websocket (rfc 7718 or OverSIP)? |
| 00:46.56 | lfamorim | and accepts credit card =p |
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| 01:30.31 | linuxmint | Anyone able fix incoming calls? Message says âRecipients mailbox is full. You cannot leave a message. Goodbyeâ. |
| 01:30.44 | linuxmint | I have dialled *97 and *98 to check voicemail is empty, and the asterisk voicemail says voicemail is empty? |
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| 01:37.05 | [TK]D-Fender | linuxmint, Are you ready to resume? |
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| 02:00.23 | linuxmint | Hi, bout to leave again. Will be back in 10 hours. Outgoing is not asteisk as I tried some other interstate numbers. However incoming needs to be fixed. |
| 02:00.45 | linuxmint | *asterisk |
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| 02:11.40 | nickgaw | Hi, Is there a way to read the asterisk book on line like a chapter at a time rather then always downloading it as one large pdf file? |
| 02:15.25 | [TK]D-Fender | ~book |
| 02:15.26 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 02:20.20 | nickgaw | Just so I am writing things correctly in modules.conf the autoload=yes should be on it's own line then the noload modules should be one per line like noload=alsa? |
| 02:21.22 | nickgaw | I am totally blind and am just making sure that Adobe acrobat is properly displaying the book. |
| 02:23.37 | nickgaw | compiling asterisk was not tricky at all but I did not use menuselect as the interface was somewhat tricky with the linux screen reader speakup is there a file I could edit with nano to determin what modules are compiled and then put it in the asterisk source directory then just run make if something is not configured like I wish or is there a way to get the list of modules that are to be compiled by default as I can edit xml files ok? |
| 02:25.29 | cresl1n | yes |
| 02:25.38 | cresl1n | all directives should be newline separated |
| 02:27.08 | nickgaw | ok just wanted to make sure I was reading it properly. |
| 02:27.21 | nickgaw | I was so that is good. |
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| 02:32.25 | nickgaw | ok that was good I was just making sure I had it right. Is there a reason that the address www.it-ebook.info is printed on I believe every page? |
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| 03:25.33 | monsterco | why don't I see asterisk verbosity level when I enter asterisk CLI using "asterisk -r" on Asterisk 13? It used to show right there on v1.8 and was very convenient |
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| 09:17.16 | kchehab | what can cause the in voice delay and choppy voice when record command enabled for outdoiing call , removing record command will let everything normal |
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| 09:18.38 | Rasputin3711 | version? |
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| 09:53.00 | cjensen | exit |
| 09:53.04 | cjensen | help |
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| 11:51.46 | modesto916 | Hi everyone. My question is kind of off-topic, but I don't know a better channel I can ask this. |
| 11:53.14 | modesto916 | A prospect customer has two PSTN numbers which he migrated to a GSM carrier. He wants to be able to have more than one customer on each line at a time. |
| 11:54.18 | modesto916 | Droping the numbers and getting an E1 is not an option since the only carrier availabe will not be selling E1 for some time. |
| 11:54.56 | modesto916 | I just tested the GSM busy forward feature on a GOIP gateway and noticed it seems to work. Do you guys know if it is reliable? |
| 11:57.22 | modesto916 | If it is in fact reliable, I'm going to recommend him to get two more lines, one to back each of his main numbers in case they are busy. |
| 11:58.46 | jonaskoeritz | forward on busy on a GSM Network will work reliably if the subscriber is stational i think |
| 11:59.14 | jonaskoeritz | there will be no race coniditions for handovers and stuff when the subscriber is always in the same cell. |
| 12:02.01 | modesto916 | In this case the sim card would be installed in a fixed location |
| 12:02.26 | modesto916 | The only problem I foresee is that the client will not be able to see the real caller id |
| 12:02.43 | modesto916 | But I don't know if it is a problem to this customer |
| 12:02.51 | modesto916 | I'll ask him |
| 12:03.53 | jonaskoeritz | yes that will be the case. Have you tried if you can see a call waiting for one line? |
| 12:09.26 | modesto916 | No, I haven't tried that. I can check now |
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| 12:13.04 | jonaskoeritz | you may get callerid info from the call-waiting, then redirect the caller and overwrite the callerid |
| 12:16.41 | modesto916 | Is it possible to overwrite the callerid sent over the gsm network? |
| 12:17.07 | jonaskoeritz | i mean if you receive the second (redirected) call you could overwrite it in asterisk |
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| 12:34.45 | modesto916 | The gateway has a hunt group feature. But I can only have one huntgroup per gateway. As the gateway is shared among different clients it would not be viable. |
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| 12:36.23 | jonaskoeritz | okay. THe client will have to accept missing callerid then. You could indicate this for the ringing phone by sending a specific id like "overflow call, number not available" or something like that. |
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| 13:29.05 | prauat | hi got problem with problem with bridging, modules are loaded, asterisk detects not nat,directmedia is set but asterisk is still using simple bridge not switching to native |
| 13:29.58 | [TK]D-Fender | Show us the call |
| 13:31.00 | file | if you turn up core debug it will tell you why not as well |
| 13:31.22 | deepa | is there any maintained webinterface for configuration? |
| 13:32.04 | prauat | realtime database configuration + some files |
| 13:34.38 | [TK]D-Fender | deepa, Thre are several GUI's for *, some free, some paid |
| 13:34.47 | prauat | http://pastie.org/10869221 |
| 13:35.10 | [TK]D-Fender | fraYou aren't showing basic verbose in there |
| 13:35.17 | [TK]D-Fender | core set verbose 10 |
| 13:35.19 | deepa | I'm currently just doing this as a hobby, so if there's a page listing free ones, preferrably open-source ones that'd be awesome |
| 13:35.20 | [TK]D-Fender | show a new call |
| 13:35.31 | [TK]D-Fender | deepa, freepbx.corg |
| 13:35.36 | [TK]D-Fender | org |
| 13:35.41 | [TK]D-Fender | the only one worth looking at really |
| 13:36.21 | [TK]D-Fender | It completely takes over configuring your system and has limited options for you to do custom things |
| 13:37.55 | file | just like the North Korean leadership! |
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| 13:38.41 | deepa | ah, but I'm a developer and doing custom things is my end-goal |
| 13:39.02 | deepa | Is there any other alternative? I'm not overly fond of the having to add php to this server |
| 13:39.38 | [TK]D-Fender | That's what pretty much all of them are written in |
| 13:39.49 | [TK]D-Fender | they are all web-based GUI's to build the configs |
| 13:40.12 | [TK]D-Fender | "how custom" is the first thing you'll need to clarify |
| 13:41.07 | [TK]D-Fender | If you're a dev you shouldn't need a GUI and can just build your own system your own way. |
| 13:41.56 | deepa | I don't *need* a GUI, it's just one of those things that'd be nice to have when starting out |
| 13:42.30 | [TK]D-Fender | You should start about having a clearer picture of the end results you want to see if it's worth it at all or not |
| 13:43.33 | deepa | That'd take the fun out of just messing around with software |
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| 15:54.28 | linuxmint | k, ready. |
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| 15:55.09 | [TK]D-Fender | For more than 7 minutes this time? |
| 15:55.16 | linuxmint | :) |
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| 16:17.26 | jrun | is accountcode available in 13.9.1 with pjsip? |
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| 18:36.53 | dan_j | Anyone here based in Manchester, UK? I need someone to cover annual leave on the Monday 13th June for the day. Please PM me your details. |
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| 18:52.01 | dan_j | Or if you are based in Israel |
| 18:52.12 | cresl1n | cover annual leave? |
| 18:53.11 | dan_j | cresl1n: I'm not available on Monday and need someone who is able to deal with asterisk issues. |
| 18:54.39 | drmessano | IRC is your backup plan? |
| 18:55.29 | dan_j | Just an option. |
| 18:55.54 | drmessano | What do you do when you go on vacation? |
| 18:56.25 | dan_j | What's that? |
| 18:56.39 | drmessano | That doesnt make sense |
| 18:58.40 | dan_j | How does it not make sense? At the moment, my business is large enough for me to worry about but too small to employ a full time asterisk engineer. |
| 19:00.04 | drmessano | Youre are not only doing yourself a disservice, but doing your customers a disservice by not allocating a sensible amount of time off for yourself. |
| 19:00.20 | cresl1n | customers don't believe in time off :-) |
| 19:00.26 | cresl1n | unless its their own |
| 19:00.37 | drmessano | No, they dont care.. thats not the concern |
| 19:01.10 | drmessano | Without time off, you're less effective over time. |
| 19:01.16 | dan_j | drmessano: you are right. i should have this situation properly sorted in the next few months i hope. |
| 19:01.49 | drmessano | Who wants to work with someone overworked, angry, and lacking in their ability to properly service them? No one |
| 19:02.03 | drmessano | So yeah, good idea |
| 19:02.24 | drmessano | At least make some friends that can get your back for a few days here and there |
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| 19:03.54 | MajesticFudgie | What command should I be looking up to add to an extension to allow a user to enter a number followed by a # but after a time limit it continues the current extension |
| 19:04.28 | MajesticFudgie | Essentially an extension menu, if no extension is entered it will jump to a designated extension |
| 19:06.15 | [TK]D-Fender | "core show application read" |
| 19:10.23 | MajesticFudgie | o.o |
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| 19:20.26 | hdon | hi all :) i'm using a legacy freepbx/asterisk/isymphony system. occasionally my calls are interrupted by, apparently, being called by myself. here is an example of what's in my asteriskccdrdb.cdr table: https://pastebin.mozilla.org/8875889 the channel column shows SIP/1802-<hex serial> for every regular call, but not for these mystery calls that appear to be originating at my own extension. instead these mystery calls have a channel of |
| 19:20.26 | hdon | "Local/1802" -- what does this say about the call? has anyone else seen this behavior? |
| 19:21.51 | *** join/#asterisk jonah (~jonah@5751acb3.skybroadband.com) |
| 19:24.18 | [TK]D-Fender | Says virtually nothing |
| 19:24.38 | [TK]D-Fender | if that is a single call it coudl a hunt groupd circling raround throughit, or CDR's created by a queue,etc |
| 19:25.55 | *** join/#asterisk KerioMorgan (~Adium@gw-us.kerio.com) |
| 19:30.15 | *** join/#asterisk tristero (~al.f.zero@unaffiliated/transfinite) |
| 19:30.22 | jonah | hi I wondered if anyone can help? I just wanted to set up something extremely basic where asterisk answers the call, plays a message, records the voicemail and then emails me it. I now have this working but it's still giving me a few problems. The phone answers, the message plays but then further generic messages play before the message is recorded. How do you stop these other messages and just go straight to the beep and message recording? |
| 19:30.55 | [TK]D-Fender | "core show application voicemail" <- |
| 19:32.50 | jonah | [TK]D-Fender: ah so I knock the ,u off the end in extensions.conf?? thanks |
| 19:33.03 | *** join/#asterisk mirela666 (~mirkob@52D9ADEB.cm-11-1c.dynamic.ziggo.nl) |
| 19:34.39 | jonah | [TK]D-Fender: can I also just replace that ,u with the ,s to skip the playback instructions? or is that for something else? |
| 19:35.07 | [TK]D-Fender | Do you want the unavailable message played back? |
| 19:35.12 | [TK]D-Fender | do you want the instructions? |
| 19:35.15 | [TK]D-Fender | It's all up to you |
| 19:35.47 | jonah | [TK]D-Fender: well not really as I already say leave a message after the beep in the background sound initially before it forwards to the mailbox... |
| 19:36.49 | [TK]D-Fender | It isn't "background" if it leads to a VM box. |
| 19:37.13 | [TK]D-Fender | But if everything you want to hear is in a sound file.. then yuo can play it back and tell the VM box to do nothing more than beep |
| 19:37.15 | jonah | [TK]D-Fender: sorry just it calls it that in extensions.conf, but no it's just my main message sorta thing |
| 19:37.33 | jonah | [TK]D-Fender: yeah that's what I'm after |
| 19:38.05 | jonah | [TK]D-Fender: so is it just exten => s,n,Voicemail(1234,s) |
| 19:38.16 | [TK]D-Fender | Go try |
| 19:38.26 | jonah | [TK]D-Fender: sure thank you |
| 19:43.06 | jonah | [TK]D-Fender: great that works but it seems to take ages for the beep... there is maybe a 5 second wait between the end of the message and then beep. but i've only put after my sound file: exten => s,n,WaitExten(0) |
| 19:43.19 | jonah | [TK]D-Fender: so i thought with that it would just come straight in with the beep really |
| 19:43.28 | [TK]D-Fender | Why are you using waitexten? |
| 19:43.37 | [TK]D-Fender | And why would you tell it 0? |
| 19:43.54 | [TK]D-Fender | If you don't want extr delays then why are you using those? |
| 19:44.15 | [TK]D-Fender | that's explicitly to get user entry |
| 19:44.33 | jonah | [TK]D-Fender: so can I just omit that line completely then? here is my pastebin: http://pastebin.com/X5iVzviQ |
| 19:45.24 | [TK]D-Fender | So you want them to be able to dial 8500 when that message plays to check VM? |
| 19:46.08 | jonah | [TK]D-Fender: no not really. I just wanted them to hear the message and then leave a voicemail and then it email me. I don't need any dial options at all. maybe that dialplan can be stripped down a little? |
| 19:46.29 | [TK]D-Fender | You have an IVR in there effectively.... |
| 19:46.38 | [TK]D-Fender | Yes don't see too aware of that fact |
| 19:46.43 | [TK]D-Fender | seem* |
| 19:47.16 | jonah | [TK]D-Fender: no, I wasn't fully aware. I didn't know which options have to stay in there or what I can safely take out so it still works |
| 19:47.48 | [TK]D-Fender | well that is built as an IVR.... doesn't sound like what you want. |
| 19:48.19 | [TK]D-Fender | What you asked about might aw well be using that other recording as a VM message and just go into the box and play it. |
| 19:48.34 | [TK]D-Fender | instead of the "play the sound then hit the VM box" as 2 separate steps |
| 19:49.48 | jonah | [TK]D-Fender: but how do i go straight in like that? what is the most simple dialplan to just achieve this, also I have my custom recording but do I need to then put it somewhere else instead of just in /var/lib/asterisk/sounds/en |
| 19:50.28 | [TK]D-Fender | You could put it in the VM folder renamed appropriately and just call Voicemail. |
| 19:52.13 | jonah | [TK]D-Fender: would you mind doing a pastebin of what [hy] would look like in extensions.conf? I can't find many guides about this |
| 19:52.55 | [TK]D-Fender | just call voicemail |
| 19:52.58 | *** join/#asterisk mirela666 (~mirkob@89.184.168.160) |
| 19:53.02 | jonah | [TK]D-Fender: or can I just remove everythign in that dialplan after exten => s,n,Hangup |
| 19:53.03 | [TK]D-Fender | after copying the file into your VM folder |
| 19:53.19 | jonah | [TK]D-Fender: and also take out the WaitExten bit too? |
| 19:53.37 | [TK]D-Fender | Why are you waiting for input? |
| 19:53.45 | [TK]D-Fender | You siad you didn't want anything to do with that |
| 19:54.01 | [TK]D-Fender | So take the bits out that have nothing to do with what you are trying to do |
| 19:54.21 | jonah | [TK]D-Fender: ok so I just have that line exten => s,n,Voicemail(1234,s) and then the hang up bit after that and that's it? |
| 19:54.36 | [TK]D-Fender | Give it a try |
| 19:54.40 | jonah | [TK]D-Fender: ok thanks |
| 19:56.45 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1:bd5a:3bd7:a2c:cdd1) |
| 19:59.11 | jonah | [TK]D-Fender: ok I changed it to this but now a woman comes out and counts says something extension then counts 1,2,3,4 and then it beeps so it sounds pretty crazy... http://pastebin.com/btZtGiKr |
| 20:00.45 | [TK]D-Fender | That would play the default "instructions" |
| 20:00.55 | hdon | [TK]D-Fender, hunt groupd? |
| 20:01.23 | [TK]D-Fender | hdon, Or queue. or Followme,. Or something else. CDR is proof of virtually nothing |
| 20:02.25 | jonah | [TK]D-Fender: ah this now seems to work instead: http://pastebin.com/5tG4u5y0 |
| 20:02.47 | jonah | [TK]D-Fender: is that ok to do it like that? seems to work... |
| 20:02.55 | [TK]D-Fender | Now use "playback" instead of "background" |
| 20:03.14 | [TK]D-Fender | Because Background will actually listen for DTMF. |
| 20:04.19 | hdon | [TK]D-Fender, yeah i can see it doesn't tell me much... i'm comparing it against asterisk logs and a packet dump i was running on the loopback interface at the time to see if something placed the call over the AMI |
| 20:04.48 | hdon | [TK]D-Fender, mostly i was curious if this was typical of some common problem that #asterisk might know |
| 20:04.57 | hdon | [TK]D-Fender, thanks for taking a look :) |
| 20:05.00 | jonah | [TK]D-Fender: ah awesome I've changed it to that and works a treat. thanks again |
| 20:05.16 | [TK]D-Fender | jonah, You're welcome |
| 20:06.57 | jonah | [TK]D-Fender: thank you. one other thing I wondered, but it may not be possible... as I get an email of the voicemail recording can it be deleted by asterisk as I can just keep them on email rather than asterisk storing them up. currently my email says I have 9 in my mailbox etc and counts up with each email i receive. can I add something in the dialplan for this to remove them after they've been emailed? |
| 20:07.22 | [TK]D-Fender | You can tell the box to delete after sending |
| 20:07.27 | [TK]D-Fender | read the sample voicemail.conf |
| 20:09.16 | jonah | [TK]D-Fender: ok thanks. I've uncommented delete=yes in that conf file. will test it now. how do I delete all the messages already stored to start fresh? |
| 20:09.44 | [TK]D-Fender | Go into the VM box |
| 20:09.55 | [TK]D-Fender | "core show application voicemailmain" <------ |
| 20:12.09 | syadnom | guys, I'm trying to mod something to get hints into a conference room w/ asterisk realtime. what column is needed to put the hint text in? |
| 20:12.20 | jonah | [TK]D-Fender: I've typed that help guide but it doens't mention how to empty the mailbox? |
| 20:12.44 | [TK]D-Fender | jonah, You go INTO it and follow the prompts and delete them just like any other voicemail system on the planet |
| 20:13.10 | syadnom | or does this actually go into the dialplan? |
| 20:13.32 | jonah | [TK]D-Fender: does that mean I need to put it in the dialplan somehow so I can dial into it though? |
| 20:13.41 | [TK]D-Fender | yes |
| 20:14.12 | jonah | [TK]D-Fender: I just wanted to delete all the messages from in asterisk cli? |
| 20:15.37 | [TK]D-Fender | There is no method from * CLI |
| 20:15.48 | [TK]D-Fender | you can kill the FILES in the vm folder directly however |
| 20:23.58 | syadnom | where the heck are dialplans even stored in realtime? |
| 20:24.01 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-85-237.hsd1.or.comcast.net) |
| 20:30.46 | hdon | syadnom, they could be in a database server |
| 20:30.57 | hdon | dunno what "realtime" is i'm a newb |
| 20:31.13 | hdon | but i heard, and it stands to reason, that there are asterisks that use a database server |
| 20:35.04 | hdon | i have some software communicating over AMI that is tagging its actionid with #AJ_ORIGINATE_<number> anyone know what software does this? |
| 20:35.51 | hdon | a google tells me about "asterisk java" which i guess is some library for working with asterisk in java... i guess that would be isymphonyserver probably then |
| 20:36.45 | hdon | surely enough, iSymphonyServer's java process has open a jar called asterisk-java-1.0.0-m2-i9.jar |
| 20:36.50 | hdon | so, that's probably it... |
| 20:54.25 | *** join/#asterisk p1ke (pike@patteri.org) |
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| 21:04.50 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 21:48.18 | *** join/#asterisk Kalavera (~Kalavera@aquiles.novelix.com.pe) |
| 21:50.47 | Kalavera | hey guys , i am facing this issue. I have two sip trunks one goes through the firewall and connect to a sip provider , this works fine. the second trunk is for evaluation purposes and it goes through a secondary router. communication for the evaluation trunk goes throught the secondary router but the other end asterisk deny it because it is presenting with the ip set up to extip |
| 21:51.11 | Kalavera | is there a way to workaround this? |
| 21:53.04 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 21:57.12 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:57.34 | Kalavera | hey guys , i am facing this issue. I have two sip trunks one goes through the firewall and connect to a sip provider , this works fine. the second trunk is for evaluation purposes and it goes through a secondary router. communication for the evaluation trunk goes throught the secondary router but the other end asterisk deny it because it is presenting with the ip set up to extip |
| 21:57.40 | Kalavera | is there a way to workaround this? |
| 22:01.45 | [TK]D-Fender | If you want the SIP packets contact info to match the IP used by the secondary route then you will either have to use a separate channel driver for that trunk, or a second transport defined in pjsip |
| 22:19.30 | *** join/#asterisk _AxS_otr (~axs@gentoo/developer/axs) |
| 22:27.54 | *** join/#asterisk _AxS_otr (~axs@gentoo/developer/axs) |
| 22:34.46 | hdon | hi all :) is it normal to see "exited non-zero" a lot in asterisk logs? to me this means something went wrong.. |
| 22:39.33 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-ieyrcvglsphxfbce) |
| 22:39.33 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 22:43.56 | [TK]D-Fender | Yes |
| 22:53.19 | MajesticFudgie | I got hold of a Sipura SPA3000 |
| 22:53.26 | MajesticFudgie | So far I'm having headaches. |
| 22:53.38 | MajesticFudgie | It clicks every so often and the lights flicker |
| 22:53.50 | MajesticFudgie | Is that something its supposed to do |
| 22:53.57 | scv | sounds about right |
| 22:54.09 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 22:54.26 | [TK]D-Fender | Any context to add? |
| 22:55.13 | MajesticFudgie | Nope |
| 22:55.15 | MajesticFudgie | Plugged it in |
| 22:55.17 | MajesticFudgie | and it clicks |
| 22:55.41 | MajesticFudgie | Not even been able to get into any form of admin controls either via a phone connected or via web |
| 22:55.58 | MajesticFudgie | It doesn't even show up on the router as connected |
| 22:56.16 | MajesticFudgie | It did a minute ago for the first time, but I couldn't get anything working and its disappeared again |
| 22:58.56 | [TK]D-Fender | Clicks audible when you're NOT on the phone? |
| 23:00.24 | scv | like a relay? |
| 23:01.40 | MajesticFudgie | yes |
| 23:01.44 | MajesticFudgie | Sounds just like a relay |
| 23:01.53 | MajesticFudgie | I picked up the phone and attempted a random dial |
| 23:02.04 | MajesticFudgie | the phone plays the dial tone, then I get silence |
| 23:02.19 | *** join/#asterisk aness (~aness@cm-84.215.174.224.getinternet.no) |
| 23:02.20 | MajesticFudgie | It clicks, I get a constant tone, clicks again, back to silence |
| 23:02.38 | [TK]D-Fender | Do you mean you hear the DTMF, or do you hear a DIAL TONE before starting to dial? |
| 23:02.48 | [TK]D-Fender | You also need to be clear about which port you're referring to |
| 23:02.48 | MajesticFudgie | DTMF |
| 23:03.01 | MajesticFudgie | The phone is in the Phone port, Line is in the Line port |
| 23:03.03 | [TK]D-Fender | If you don't hear dialtone on the FXS port then it is not registered |
| 23:03.22 | [TK]D-Fender | Because it won't give you tone until then |
| 23:03.32 | MajesticFudgie | If I mess with the voltage, I'm able to make a call through my provider |
| 23:03.50 | MajesticFudgie | The actual telephone provider for my home |
| 23:03.57 | MajesticFudgie | Even when the device is off |
| 23:04.13 | [TK]D-Fender | that is a passthough relay |
| 23:04.16 | MajesticFudgie | ah |
| 23:04.21 | [TK]D-Fender | Which is a feature it has |
| 23:04.30 | MajesticFudgie | So if the device is off it will pass from one port to the other? |
| 23:04.36 | [TK]D-Fender | if you tell it to |
| 23:04.43 | MajesticFudgie | Well idk how to not tell it to :P |
| 23:04.54 | [TK]D-Fender | MAGIC SETTINGS in the interface |
| 23:04.59 | MajesticFudgie | ah |
| 23:05.15 | MajesticFudgie | So, if it's clicking even when I'm not making a call. What does this mean? |
| 23:05.15 | [TK]D-Fender | #veryfinemanuals |
| 23:05.30 | [TK]D-Fender | probably improper line settings |
| 23:05.55 | MajesticFudgie | What should I do? |
| 23:06.04 | MajesticFudgie | Bearing in mind I'm having trouble accessing any form of admin interface |
| 23:06.06 | [TK]D-Fender | Go look? |
| 23:06.11 | [TK]D-Fender | Read manual? |
| 23:06.15 | [TK]D-Fender | Buy low? |
| 23:06.17 | MajesticFudgie | Manual is of no use |
| 23:06.18 | [TK]D-Fender | Sell high? |
| 23:06.25 | [TK]D-Fender | Manual is of plenty use |
| 23:06.37 | MajesticFudgie | It says to dial **** to enter the IVR menu for config etc |
| 23:06.43 | MajesticFudgie | But nothing happens |
| 23:06.55 | [TK]D-Fender | That could also be disabled |
| 23:07.00 | [TK]D-Fender | Gotten into the GUI yet? |
| 23:07.15 | MajesticFudgie | Nope |
| 23:07.24 | MajesticFudgie | Idk if the previous owner reset it |
| 23:07.25 | [TK]D-Fender | So you haven't configured anything yet? |
| 23:07.28 | MajesticFudgie | I assume it'll be on port 80 |
| 23:07.28 | MajesticFudgie | nope |
| 23:07.39 | MajesticFudgie | When I get to that stage I'll be fine lol |
| 23:07.48 | [TK]D-Fender | So your dialing attempts... were on an unconfigured (for your server) device. |
| 23:08.05 | [TK]D-Fender | Which should start with an assumption of failure |
| 23:08.09 | [TK]D-Fender | Which is where we are |
| 23:08.15 | MajesticFudgie | **** is part of the device |
| 23:08.25 | MajesticFudgie | You dial it to enter a config menu |
| 23:08.29 | MajesticFudgie | Other than the GUI |
| 23:08.34 | [TK]D-Fender | Unless it's DISABLED |
| 23:08.47 | [TK]D-Fender | Which is something that can be set |
| 23:09.01 | [TK]D-Fender | As they can also be provisioned off other means |
| 23:09.03 | MajesticFudgie | True |
| 23:09.06 | MajesticFudgie | Yeah |
| 23:09.33 | MajesticFudgie | I wonder how you're supposed to fix issues with them if you cant get into it lol |
| 23:09.51 | [TK]D-Fender | Manual <- |
| 23:09.55 | [TK]D-Fender | Factory Reset |
| 23:10.05 | MajesticFudgie | You can only reset via an admin menu |
| 23:10.12 | MajesticFudgie | GUI / IVR |
| 23:10.24 | [TK]D-Fender | Proved that it's pulled an address yet? |
| 23:10.31 | MajesticFudgie | IP? |
| 23:10.34 | [TK]D-Fender | Done a full port scan? |
| 23:10.49 | [TK]D-Fender | Attempted a series of the DTMF functions? |
| 23:10.50 | MajesticFudgie | For a brief minute it appeared with an IP under my routers devices |
| 23:10.56 | MajesticFudgie | But not long enough for me to do much |
| 23:11.07 | MajesticFudgie | Other than try the obvious port 80, 8000 and 8080 |
| 23:11.17 | [TK]D-Fender | how does an ip "disappear" instantly? |
| 23:11.21 | MajesticFudgie | By default it should be 80 according to its manual |
| 23:11.27 | MajesticFudgie | No idea, but It managed it |
| 23:11.33 | [TK]D-Fender | Go scan |
| 23:11.36 | MajesticFudgie | it was showing for all of 5minutes, not even that |
| 23:11.56 | [TK]D-Fender | And I don't mean the Battleship Whack-a-mole method |
| 23:13.34 | MajesticFudgie | Running nmap now |
| 23:15.27 | MajesticFudgie | Nothing else on the network other than whats supposed to be |
| 23:15.33 | MajesticFudgie | It's completely disconnected from it |
| 23:23.28 | MajesticFudgie | "You have to reset it to factory defaults." |
| 23:23.37 | MajesticFudgie | Just noticed that in the sellers listing |
| 23:25.16 | [TK]D-Fender | That's code-words for "I bought this second-hand off eBay too and it was provider-locked and I could never get it working.... and neither will you, but that'll be YOUR problem" |
| 23:25.19 | [TK]D-Fender | #soldasis |
| 23:25.27 | MajesticFudgie | yeah |
| 23:25.35 | MajesticFudgie | Well I'll mess with it and see |
| 23:25.40 | MajesticFudgie | Otherwise not a massive loss |
| 23:25.53 | [TK]D-Fender | This is why I tell people to not to cheap-ass chumps and get screwed trying to save 5$ |
| 23:27.15 | MajesticFudgie | Well I was suggested a SPA-3000 and it was the best I could find lol |
| 23:27.43 | MajesticFudgie | the UK seems completely void of any useful hardware |
| 23:28.01 | MajesticFudgie | I have to pay a modest shipping fee for a different country to get anything |
| 23:38.23 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 23:41.10 | [TK]D-Fender | http://www.voipfone.co.uk/hardware.php |
| 23:42.00 | [TK]D-Fender | https://www.broadbandbuyer.co.uk/store/voip-sip-phones-routers-ip-pbx/ |
| 23:42.12 | [TK]D-Fender | http://www.voipstore.co.uk/acatalog/ |
| 23:42.25 | [TK]D-Fender | http://www.voipon.co.uk/ip-desk-phone-product-comparison.php |
| 23:42.31 | [TK]D-Fender | "Void" my ass |
| 23:46.44 | MajesticFudgie | Nothing you linked appears to have both an FXS and FXO port |
| 23:47.36 | *** join/#asterisk wil_syd (~wil_syd@c110-20-159-70.rivrw10.nsw.optusnet.com.au) |
| 23:47.47 | MajesticFudgie | All I want is to be able to connect a standard RJ11 phone and make calls through the PSTN Network |
| 23:51.59 | [TK]D-Fender | https://shop.voicehost.co.uk/ |
| 23:52.02 | [TK]D-Fender | http://www.voipfone.co.uk/shop.php?method=view&pid=15 |
| 23:52.13 | [TK]D-Fender | Don't be LAZY. |
| 23:52.27 | [TK]D-Fender | I know it's a stretch but they have WONDERFUL PRODUCT PACGES |
| 23:52.36 | [TK]D-Fender | Tha list producsts JUST LIKE THIS |
| 23:52.46 | [TK]D-Fender | So YES, there ARE plenty of UK vendors |
| 23:52.54 | [TK]D-Fender | So get off your ass and look. |
| 23:55.30 | *** part/#asterisk kharwell (kharwell@nat/digium/x-fpfjiparoptjkdsp) |