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| 03:03.12 | igcewieling1 | ketas: could you be using a hostname for the remote side which has more than one ip address? |
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| 06:08.03 | ChannelZ | hrrm.. just noticed something on a Red Hate system - doing an ls -la a few files have a . after the last permission (so it says -rw-r--r--. for instance) |
| 06:08.33 | ChannelZ | WTF does the . mean? |
| 06:08.57 | ChannelZ | this is kind of hard to Google |
| 06:15.56 | ChannelZ | oh.. something relating to SELinux |
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| 06:25.34 | wyoung | ChannelZ: I know what the plus means, I didn't know about the dot tough :) |
| 06:25.37 | wyoung | though* |
| 06:28.33 | ChannelZ | ls -laZ shows the "selinux context" on these as "system_u:object_r:file_t:s0" which means nothing to me as I don't normally run selinux |
| 06:28.48 | ChannelZ | shrugs |
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| 07:06.29 | dakudos | ChannelZ : ls -Z will show SeLinux Contexts for files. These contexts (as I understand it) reference security related information. |
| 07:08.18 | dakudos | I believe each bit of information is seperated by : ... system_u is the user/Login Name object_r is a role type, file_t should be the default type for files and s0 is the lowest sensitivety level. |
| 07:09.33 | dakudos | You would use this information to make access control decisions |
| 07:11.10 | dakudos | perhaps that information is not really that helpful, depending on the answers you are looking for. :) |
| 07:11.52 | dakudos | I dont generally run SeLinux myself... my apologies if that was not helpful enough. |
| 07:17.19 | ChannelZ | I don't either so I don't understand the finer points. But I guess I don't really care, selinux isn't turned on for this system anyway, but that's the first time I noticed the . at the end of permissions on something |
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| 07:25.25 | dakudos | the dot you speak of, I think is a place holder to represent an empty ACL. |
| 07:27.36 | seik0 | Now AMI answers within 30ms, yesterday it was 4-6 sec, still can't figure out the reason |
| 07:28.17 | dakudos | oh wait, not the character probably indicates a file with an SELinux |
| 07:28.17 | dakudos | <PROTECTED> |
| 07:28.23 | *** join/#asterisk moe` (~moe@node-be028882.scarlet.cw) |
| 07:28.31 | moe` | hey kids, its me again |
| 07:28.48 | seik0 | asterisk often uses curl and odbc queries, maybe under load some locking issue |
| 07:30.11 | moe` | I'ma wondering, how far a throw is it to get asterisk do start doing voicemail? |
| 07:30.32 | moe` | I have two SIP trunks, Skype and FlowRoute |
| 07:30.43 | moe` | so inboud I have no problem |
| 07:30.47 | robmal | There's an app fro that! |
| 07:31.06 | moe` | and outbound I set prefix 9 for skype and 7 for flowroute |
| 07:31.18 | robmal | You really are from Curacao? |
| 07:31.30 | seik0 | moe, continue |
| 07:31.52 | moe` | robmal : yes, I am in Curacao |
| 07:32.33 | moe` | I'm just wondering what voicemail.conf means here |
| 07:32.52 | moe` | robmal: unixninja.net |
| 07:32.55 | robmal | Is your tap water 30% alcohol and blue? |
| 07:33.01 | seik0 | can't see connections between outbound prefixes and voicemail |
| 07:33.07 | ChannelZ | voicemail is easy-peasy |
| 07:33.11 | moe` | seik0 : correct |
| 07:33.26 | robmal | https://wiki.asterisk.org/wiki/display/AST/Configuring+Voice+Mail+Boxes |
| 07:33.27 | moe` | tell me the magic ChannelZ |
| 07:33.40 | ChannelZ | there's little magic. Config a mailbox in voicemail.conf |
| 07:34.11 | moe` | ok, lemme have a looksy |
| 07:34.18 | ChannelZ | then in your dialplan, use a timeout on your Dial followed by a call to VoiceMail(box@context) or whatever |
| 07:34.35 | robmal | VMs can wait, is your tap water alcohol? |
| 07:34.57 | moe` | no it's beer |
| 07:34.57 | ChannelZ | Then make another extension for your 'internal' voicemail pickup (500 seems to be a favorite) that calls VoiceMailMain |
| 07:35.05 | robmal | Oh noes, even better. |
| 07:35.06 | moe` | we drink a lot of beer on this island |
| 07:35.49 | moe` | So ChannelZ ... I'm left with a default voicemail.conf |
| 07:36.01 | robmal | Check the link i posted to asterisk wiki, it's all there. |
| 07:36.24 | ChannelZ | [default] |
| 07:36.30 | ChannelZ | 100 => 1234,Your Name |
| 07:36.36 | moe` | I've molested sip.conf and extensions.conf to the n'th degree |
| 07:36.42 | ChannelZ | makes mailbox 100 with a password 1234. |
| 07:36.53 | moe` | really? |
| 07:36.56 | moe` | that simple eh |
| 07:37.02 | ChannelZ | the [general] section is necessary to set format and things |
| 07:37.12 | moe` | so no extensions.conf to make it go to mailbox? |
| 07:37.51 | ChannelZ | Yes, I said that. Put a timeout on your Dial that calls your phone, so if no one answers it goes on with the dialplan and calls VoiceMail(100@default) for example |
| 07:38.27 | moe` | ok so dialplan needs an adjustment |
| 07:38.32 | ChannelZ | anyway the [general] section from the default voicemail.conf is fine to start with |
| 07:38.39 | moe` | so where does it store stuff? |
| 07:38.45 | moe` | I need to tell it where |
| 07:39.05 | ChannelZ | it does it its self. /var/spool/asterisk or something, I forget |
| 07:39.53 | ChannelZ | read the default voicemail.conf, it documents everything |
| 07:40.08 | moe` | I see that, compiled asterisk from source but I see it's writable for the daemon |
| 07:43.21 | moe` | so what's the magic, for example I have SIP trunks to Skype and flowroute and now I want inbound voicemail for a flowroute provided number. I have extensions.conf setup to forward the inbound to the SIP client... but extensions.conf must be (so I guess) setup to failover to voicemail after a timeout |
| 07:43.47 | ChannelZ | I've explained twice now |
| 07:44.04 | moe` | sorry, I've had a few beers |
| 07:44.12 | moe` | so humor me |
| 07:44.17 | ChannelZ | then programming is probably a poor choice. |
| 07:44.30 | ChannelZ | Scroll back. I can only type the exact same thing so many times |
| 07:44.45 | moe` | sometimes I crank out my best C code or shell scripts when I'm hammered :) |
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| 08:05.44 | moe` | man, I am stuggling |
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| 08:06.13 | moe` | exten=>15065000060,1,Dial(${FRANCIS_DEVICES},60) |
| 08:06.13 | moe` | exten=>15065000060,2,VoiceMail(15065000060) |
| 08:06.20 | moe` | is that sane? |
| 08:07.08 | moe` | if so, what do I put in voicemail.conf? |
| 08:08.13 | moe` | Sorry ChannelZ I am brain dead |
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| 08:18.52 | xpheres | hello |
| 08:19.03 | xpheres | I'm trying to make tls encryption work |
| 08:19.28 | xpheres | however, when I try to log in from an extension with encryption on in the server and extension I'm not able to log in |
| 08:19.48 | xpheres | I receive transport failure at the softphone |
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| 08:30.35 | moe` | xpheres : I hear that, been there, don't have an answer. |
| 08:31.06 | moe` | for UDP connetions it's dtls, but still, no joy. |
| 08:31.48 | xpheres | I'm not sure I understand |
| 08:31.57 | xpheres | you mean it does not work properly zet? |
| 08:31.59 | xpheres | yet? |
| 08:35.48 | xpheres | peer '103' HAS NOT USED (OR SWITCHED TO) TLS in favor of 'UDP' (but this was allowed in sip.conf)! |
| 08:35.50 | xpheres | is this normal? |
| 08:35.59 | xpheres | can TLS work over udp transport? |
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| 08:51.01 | moe` | aw man I'm missing something basic |
| 08:51.12 | moe` | for voicemail |
| 08:51.30 | moe` | I'm an idiot, wtf am I doing wrong |
| 08:51.53 | xpheres | ask |
| 08:52.47 | moe` | yeah I did, ChannelZ told me rightly I was being dumb |
| 08:53.00 | moe` | no fault of his |
| 08:53.14 | moe` | I'm just missing the magic |
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| 09:10.04 | WIMPy | There is no magic. And ChannelZ already told you everythign else. |
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| 09:17.39 | xpheres | everytime I make a call with enabled encryption I see the message network busy |
| 09:20.05 | xpheres | Inband DTMF is not supported on codec speex. Use RFC2833 |
| 09:20.08 | xpheres | maybe that's the reason? |
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| 09:56.11 | ketas | igcewieling1: hmm? |
| 09:58.50 | ketas | igcewieling1: there are srv records... but only highest priority server (of two) ever seems to be used |
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| 10:06.06 | *** join/#asterisk kedare (~Kedare@104.ip-51-254-128.eu) |
| 10:06.08 | kedare | Hi all o/ |
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| 10:07.10 | kedare | I have a little issue, I am using AMI and I want to use the Originate command but I get a Permission denied... It's quite weird because my permissions looks good in "manager show user" : system,call,log,verbose,command,agent,user,originate,all |
| 10:07.23 | kedare | (In both read and write) |
| 10:07.29 | kedare | What could be the cause ? |
| 10:07.55 | kedare | (I can read call without issues) |
| 10:10.13 | WIMPy | Did you reload the config and reconnect? |
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| 10:12.04 | kedare | WIMPy: Yes, I did "manager reload" |
| 10:12.18 | kedare | And the correct permission appearon "show manager user xxx" |
| 10:14.04 | WIMPy | Do you make a new connection every time? The permissions aren't changed on persistant connections. |
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| 10:16.44 | kedare | Yes :) |
| 10:17.08 | kedare | It's on a standalone script for now that I run at every time (for testing) |
| 10:18.22 | kedare | Could the permission denied be cause by a wrong parameter in the command ? |
| 10:18.23 | kedare | http://pastebin.com/HJMegt91 |
| 10:18.48 | wasanzy | Hello am getting this error when trying to do OBD: http://pastebin.com/93DRjc7S |
| 10:20.54 | WIMPy | kedare@ That does not look like a valid channel name. |
| 10:21.38 | WIMPy | wasanzy: I dind't know Asterisk was compatible to On Board Diagnostics. :-) |
| 10:22.18 | WIMPy | wasanzy: Looks like you tried to call an invalid/non-existant number. |
| 10:22.47 | WIMPy | Hmmm. |
| 10:23.01 | wasanzy | The number I called exist cus when I call same number with my handset, it goes through |
| 10:23.09 | WIMPy | Actually the "reason" at that point might have another meaning. |
| 10:23.57 | WIMPy | But the "forbidden" is an obvious one. |
| 10:28.54 | wasanzy | ok |
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| 11:05.57 | kedare | WIMPy: What do I have to use a channel name ? Just SIP/202-abcdef with a unique id ? |
| 11:06.17 | WIMPy | The same you would use in Dial(). |
| 11:06.26 | kedare | SIP/202 for example so ? |
| 11:07.00 | WIMPy | If that's a configured peer name, yes. |
| 11:07.06 | kedare | Hmm same error with this |
| 11:08.02 | WIMPy | Might help to see what's actuially sent to AMI. |
| 11:08.18 | kedare | http://pastebin.com/dtKX7adL |
| 11:08.50 | kedare | Maybe I there is someway to get more info from the asterisk CLI ? |
| 11:09.16 | WIMPy | increase verbose and/or debug levels. |
| 11:09.36 | WIMPy | What kind of debug is that? |
| 11:10.16 | WIMPy | But that looks like a permission thing. Are you using the correct user to log in? |
| 11:10.48 | kedare | Sure |
| 11:11.00 | kedare | I use the same user for something else and I don't have any issue (for read only stuff) |
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| 11:13.04 | kedare | But I don't see the connect evens on the cli (Just started with asterisk -rvvvvv) |
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| 11:17.47 | cervajs2 | hi, is it possible log asterisk call-id to cdr? what's best practice? |
| 11:18.38 | WIMPy | Which call-id? |
| 11:19.57 | WIMPy | kedare: Debug 1 will give you the full communication. |
| 11:22.14 | kedare | Just "debug 1" on the CLI ? |
| 11:22.37 | WIMPy | core set debug 1 |
| 11:22.50 | WIMPy | or asterisk -rvvvd |
| 11:23.23 | kedare | Okay, I will try that when possible :) |
| 11:23.27 | kedare | Thanks |
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| 11:29.44 | stefan27 | WIMPy you're a good person, you're always answering people's questions here |
| 11:30.19 | WIMPy | If I did it *always*, I'd have some serious issues. |
| 11:30.29 | WIMPy | Well, I guess I have anyway... |
| 11:30.45 | stefan27 | youre kind of like file in asterisk-dev |
| 11:31.03 | file | moo |
| 11:32.02 | WIMPy | Or here as well? |
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| 11:35.36 | WIMPy | Let's combine the topics AMI and -dev... |
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| 11:41.07 | cervajs2 | this call-id https://wiki.asterisk.org/wiki/display/AST/Call+Identifier+Logging not SIP Call-ID |
| 11:42.13 | cervajs2 | i want debug problematic calls. first i'll search problematic call in CDR. then i'll check /var/log/messages for dialplan problems by call-id |
| 11:44.28 | kedare | WIMPy: Hmm I see nothing in the logs with core debug at 1 |
| 11:45.05 | WIMPy | Do you have debug enabled for the console in logger.conf? |
| 11:46.07 | kedare | Hmm I just have this line : full => notice,warning,error,debug,verbose |
| 11:47.29 | WIMPy | You should have debug in console =>. |
| 11:48.49 | kedare | Okay I uncommented the line with console and now I see more info |
| 11:49.49 | kedare | But the only thing related to manager is manager.c:4719 process_message: Running action 'Login' |
| 11:50.06 | kedare | All the rest is audiohook and chan_sip |
| 11:52.51 | WIMPy | Sorry, looks like you need debug 3 or more. |
| 11:55.22 | kedare | I have a little idea, haha |
| 11:55.37 | kedare | I think the login action is Async so it try to run the action before being logged in |
| 11:56.22 | WIMPy | Yes, you have to wait for resopnes to your actions. |
| 11:57.00 | WIMPy | But it looked like you're using some lib there. Doesn't it handle that for you? |
| 11:57.15 | kedare | That was the issue |
| 11:57.17 | kedare | :) |
| 11:57.25 | kedare | It does |
| 11:57.36 | kedare | But I forgot to put the code in the event that is send when the login is done |
| 11:57.43 | kedare | Sorry >< |
| 11:57.55 | WIMPy | Huh? |
| 11:58.27 | kedare | I put the Originate command just after starting to login, and not in the event that is called when the client has logged into AMI |
| 11:58.59 | kedare | Like in the samples : https://github.com/marcelog/Nami/blob/master/src/index.js#L60 |
| 11:59.05 | WIMPy | That would be THE thing I'd expect from some library to take care of for me. |
| 11:59.22 | kedare | It's async by default :) |
| 11:59.25 | kedare | JS world |
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| 11:59.43 | WIMPy | What's the point of that lib then? |
| 12:00.24 | kedare | All the libs in JS will be async |
| 12:00.35 | kedare | And I need async and event based for other stuff :) |
| 12:01.53 | WIMPy | I would have expected some queueing or blocking. Just formatting the messages shouldn't require an extra library. |
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| 12:38.20 | killermoehre | hi. question: will the repo at packages.asterisk.org ever be updated to include dahdi-linux 2.10.0 for centos6? right now the local admin is compiling this by hand and I want it to be available via the local package repo |
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| 12:58.23 | kedare | WIMPy: The purpose on this library is to have everything event-based :) |
| 12:58.50 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 12:59.13 | WIMPy | That's the very nature of AMI itself. |
| 13:02.08 | *** part/#asterisk cervajs2 (~cervenka@gatekeeper.bm.ipex.cz) |
| 13:02.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:08.06 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 13:08.21 | kedare | Hmm, is there a way to differentiate a "normal" inbound call and a call initiated by Originate (calling the caller) ? |
| 13:09.44 | kedare | I use Newchannel and Newstatus currently to track calls |
| 13:10.42 | WIMPy | Take a look at "ActionId". that should hopefully provide a link. |
| 13:10.56 | kedare | Thanks |
| 13:11.30 | WIMPy | But I haven't tried that myself. Hence the hope :-) |
| 13:12.39 | kedare | I tried to set a variable with Originate and was hoping to find it on all the events related to this call but looks like it's not working this way |
| 13:13.14 | WIMPy | No, that's a channel variable. |
| 13:14.17 | [TK]D-Fender | ActionID IIRC <- |
| 13:15.15 | kedare | But the ActionID can only be found on the Originate command and response, no ? |
| 13:15.47 | WIMPy | Doesn't the response include some other ID? |
| 13:16.26 | [TK]D-Fender | Actually, correct, it should not continue though the execution of the actual new channel AFAIK |
| 13:16.53 | kedare | Nop, the answer just include the ActionID and the status (succes) |
| 13:17.00 | kedare | Not even the Channel |
| 13:18.36 | kedare | Hmm maybe I can create a context just for this and include the one I use ? |
| 13:18.48 | kedare | To the specific one would appear in the events |
| 13:18.50 | kedare | So* |
| 13:19.03 | WIMPy | That would work. |
| 13:19.26 | *** join/#asterisk fornax (~fornax@85.183.43.221) |
| 13:20.08 | kedare | I will try that |
| 13:22.38 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:25.49 | kedare | Hmm, nop, the last context overwrite it on the Newchannel event |
| 13:26.00 | kedare | But I can just use a specific value on callerid and use it |
| 13:27.40 | killermoehre | nobody knows if packages.asterisk.org becomes updated to include dahdi-linux 2.10.X for centos6? |
| 13:30.18 | WIMPy | Overwrite? |
| 13:30.23 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 13:31.12 | kedare | WIMPy: For example I just created a context "cti-initiated" and just include "from-internal" inside, on all the Newchannel it's using from-internal context |
| 13:33.32 | killermoehre | or any reason why the open suse build service is disabled for centos? |
| 13:34.32 | file | packages are a best effort thing when resources allow, which has not been available lately |
| 13:35.51 | killermoehre | file: so you guys don't have an automated build environment? hmmm |
| 13:35.58 | file | not currently, no |
| 13:36.31 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 13:36.54 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 13:39.31 | killermoehre | file: but there is a git repo, right? |
| 13:39.37 | file | for what? |
| 13:40.37 | killermoehre | for the source the packages are build from |
| 13:41.21 | file | Asterisk is kept in git, as is DAHDi |
| 13:42.02 | file | https://gerrit.asterisk.org/ for Asterisk and http://git.asterisk.org/gitweb/ for DAHDi |
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| 13:42.13 | killermoehre | ok, thank you |
| 13:43.17 | *** part/#asterisk emdk (~emdk@pool-74-110-110-254.nrflva.fios.verizon.net) |
| 13:44.07 | killermoehre | ha, I can't access those sites, because the firewall here is blocking me -_- |
| 13:44.41 | *** join/#asterisk emdk (~emdk@pool-74-110-110-254.nrflva.fios.verizon.net) |
| 13:45.00 | file | there are also mirrors on https://github.com/asterisk?page=1 |
| 13:47.40 | killermoehre | purrfect |
| 13:47.48 | killermoehre | :3 |
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| 13:56.15 | voip | nice |
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| 14:59.26 | Kobaz | anyone have any issues with microbrowser idle refresh only loading the page once |
| 14:59.31 | Kobaz | er, for a polycom phone |
| 14:59.43 | Kobaz | so weird |
| 15:19.33 | Iamnacho | I'm trying to decide if I need a full PBX for a project: I have a legacy PBX that is not VoIP and is Analog only. I have a single remote user that I'd like to have a phone extension as VoIP. Is there such a device that would let me hook say a Sip phone directly to an "appliance" or "adpater" to convert an analog line of my non-VoIP PBX to SIP? |
| 15:20.06 | Iamnacho | I'm thinking: analog Extension -> Some Magic Device -> Internet/VPN -> SIP Phone. |
| 15:20.20 | Iamnacho | What's the "Some magic Device" really called (if it exists)? |
| 15:20.39 | WIMPy | Some ATAs also have an FXO port to do the reverse of what an ATA usually does. |
| 15:21.37 | WIMPy | Someone will probably come up with a well known Linksys device. |
| 15:21.47 | Iamnacho | I have an FXO (a grandstream HT503) that I use for another aplication to pull my POTS phone line to my asterisk server. But I've never seen inside of the HT503 where I can set settings for a sip phone to register directly to it |
| 15:22.51 | WIMPy | It might not have that option. I don't know if the Linksys has, either. |
| 15:23.02 | Iamnacho | to note, these 2 projects are completely separate and unrelated. I only bring it up because of my limited knowledge of FXO devices |
| 15:23.11 | [TK]D-Fender | The pieces are there, it's a question of if the software lets you orient it that way |
| 15:23.25 | WIMPy | I guess you wouldn't have a static host for the phone, would you? |
| 15:23.30 | [TK]D-Fender | Iamnacho: and that is the kind of device you'd use. |
| 15:23.48 | [TK]D-Fender | Iamnacho: However you might need something like * to sit inbetween. |
| 15:24.25 | Iamnacho | so to the best of both of your knoweldge, there's not a specific device that you've used that I should buy. |
| 15:25.03 | WIMPy | hasn't done anything analog for much longer than we have voip. |
| 15:25.17 | Iamnacho | WIMPy: what do you mean static host ? the remote SIP phone would be connected back to the "home office" where the legacy PBX exists via VPN, so yes to static I think. |
| 15:25.46 | WIMPy | That might be good wnough to configure that ATA. |
| 15:25.49 | Iamnacho | My suggestion was to replace the POS PBX, but that was shot down :) |
| 15:27.05 | WIMPy | FritzBoxes would definitely do the trick, at least those that still have an FXO port, but it would be cheaper to use some other decive and a Raspberry PI with Asterisk on it in between. |
| 15:28.24 | *** join/#asterisk [Outcast] (~outcast@2601:18f:200:4a4b:d9cd:8dd8:479f:3185) |
| 15:29.47 | Iamnacho | interesting. So I think I undestand that I'm not overlooking something and that it would be easier to put * in the middle to handle the conversion. |
| 15:30.22 | WIMPy | It's just a question of what you can configure on the device. |
| 15:30.52 | Iamnacho | That's good to know, I spent a few hours yesterday reading about various ATA/FXO devices and could never find the part where I could register a SIP device to it. I always saw that the FXO registered back to some other SIP Server. |
| 15:31.25 | WIMPy | If you have static IPs in your VPN, there's no need to register. |
| 15:32.13 | Iamnacho | ?? I don't undestand that comment (and this might be the piece I am missing) can you expand upon that? |
| 15:32.31 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:32.53 | WIMPy | Registration only serves the purpose to make your location known to the other end in case it changes. |
| 15:36.59 | Iamnacho | ok, so I would configure and FXO to send incoming analog calls to the static IP of my SIP Desk phone. On the SIP desk phone I would do the same, but in reverse? |
| 15:37.47 | Iamnacho | would that maybe require the SIP phone to have a dial plan setup within it? |
| 15:43.45 | WIMPy | Yes. |
| 15:44.18 | WIMPy | The dialplan is there to determine when you have finished dialling without having to hit send or the like. |
| 15:49.27 | Iamnacho | ok that explains a few thing. I think part of my problem is that I've only ever played with crappy linksys sip phones. I don't know that they allow this type of configuration |
| 15:50.09 | *** join/#asterisk dms1 (~dms1@162.207.112.26) |
| 15:50.53 | [TK]D-Fender | Your linksys phones HAVE a dialplan on them |
| 15:51.21 | [TK]D-Fender | The only phones I've seen are Grandstrreams shittier phones, and PA1688 chip based garbage |
| 15:53.32 | Iamnacho | right, I'm only asking for clarification, but the dialplan would need to say "send all calls to this IP address" correct ? |
| 15:55.05 | [TK]D-Fender | it COULD |
| 15:55.47 | WIMPy | What kind of dialplan are we talking about? |
| 15:56.05 | [TK]D-Fender | so far "choosing when to consider a number complete and passing it off" |
| 15:56.10 | *** join/#asterisk navaismo (c9747d24@gateway/web/freenode/ip.201.116.125.36) |
| 15:56.29 | Iamnacho | we are talking about the theory of how to point my non-registered SIP phone at an FXO to have it send calls there |
| 15:56.47 | WIMPy | Ok, the just started to sound like a routing table. |
| 15:57.37 | Iamnacho | I'm not entirely sure that I undestand how the non-registered part works |
| 15:58.08 | Iamnacho | I mean the theory, and it's because I have limitied knoweldge of all of this. :) |
| 15:59.50 | [TK]D-Fender | ~sipregister |
| 15:59.51 | infobot | [~sipregister] SIP registration is to tell your provider what IP address & EXTEN to send INCOMING calls to. Some ITSPs let you use a fixed address or host rather than registering. Registration is NOT normally needed to PLACE calls, as those are typically auth'ed independently. Others accept unauth'ed calls once you are registered (saves on negotiation BW). |
| 16:01.09 | WIMPy | Generelly you send calls to user@host, the advanced version is user@(the host that registered .account xxx) |
| 16:07.03 | *** join/#asterisk monsterco (~monsterco@70.49.229.188) |
| 16:09.00 | Iamnacho | WIMPy & [TK]D-Fender: Thank you both for your time and help. Though not related to *, I knew that someone here would be able to help me with this :) |
| 16:09.15 | [TK]D-Fender | Could still be |
| 16:09.31 | [TK]D-Fender | Not sure if that gateway can sanely do what you want all on its own |
| 16:09.45 | Iamnacho | I agree and understand |
| 16:10.09 | Iamnacho | Thank you :) |
| 16:16.49 | *** join/#asterisk Encrypt (~Chuck@cpc65608-newt33-2-0-cust287.19-3.cable.virginm.net) |
| 16:16.53 | Encrypt | Hi! o/ |
| 16:17.39 | Encrypt | I would like to connect my SIP client to my remote asterisk server |
| 16:18.03 | Encrypt | MY asterisk server is behind a NAT but I can forward ports on the router |
| 16:18.15 | Encrypt | Is there any solution to do that? |
| 16:18.22 | Encrypt | I've heard about IAX |
| 16:20.17 | [TK]D-Fender | Go forward the appropriate ports |
| 16:20.48 | Kobaz | Iamnacho: your magic device is a FXS ATA |
| 16:20.48 | Encrypt | [TK]D-Fender, Btw, my setup would be: |
| 16:21.22 | Kobaz | iamkorvus: prices have come waaaaay down on these things |
| 16:21.26 | Kobaz | whoops |
| 16:21.30 | Kobaz | ^ Iamnacho |
| 16:21.34 | Encrypt | Asterisk server ---> modem/router with appropriate opened ports ---> WAN ---> modem/router with NAT ---> my SIP client |
| 16:21.38 | Kobaz | Iamnacho: http://www.newegg.com/Product/Product.aspx?Item=N82E16833150140 |
| 16:21.54 | [TK]D-Fender | Encrypt: Whcih is fine |
| 16:22.32 | [TK]D-Fender | Encrypt: Forward SIP & RTP to your * server, and set it up properly to work from behind NAT where it knows what your WAN IP is, prevents reinvites, etc |
| 16:23.14 | navaismo | *whisper* and prepare for the SIP attacks |
| 16:23.19 | Kobaz | aye |
| 16:23.25 | Kobaz | sip pirates |
| 16:23.37 | Encrypt | navaismo, Is it that "severe"? |
| 16:23.44 | Encrypt | I mean worse than SSH attacks? |
| 16:23.47 | Kobaz | yeah |
| 16:23.49 | Kobaz | asterisk can't handle it |
| 16:24.00 | navaismo | can I suggest to use a VPN?? |
| 16:24.01 | Kobaz | i was seeing 100 requests/sec on a server the other day |
| 16:24.08 | Kobaz | asterisk craps out really badly when that happens |
| 16:24.20 | Encrypt | Hum (<.<) |
| 16:24.25 | Encrypt | Even with fail2ban? |
| 16:24.26 | Kobaz | i have some rate limiting rules i've been putting on various servers to prevent that |
| 16:24.29 | Kobaz | well |
| 16:24.34 | Kobaz | this is when fail2ban stops working |
| 16:24.37 | Kobaz | which is, always |
| 16:25.02 | Kobaz | fail2ban has been such a fail, i wrote my own, that has 100% uptime |
| 16:26.09 | navaismo | yealink pohones can connect to openvpn a poor implementation but still better than open the gates of the hell |
| 16:26.41 | cyford | opensip |
| 16:26.50 | Encrypt | What about using a VPN to be directly connected to the mlocal network where my asterisk server is? |
| 16:26.53 | *** join/#asterisk r00f (~r00f@av.r00f.us) |
| 16:26.58 | Kobaz | yeah |
| 16:27.00 | Kobaz | that's fine |
| 16:27.05 | Kobaz | block your public ports other than the openvpn ports |
| 16:27.06 | Encrypt | I haven"t done that though |
| 16:28.57 | cyford | u usually only allow providers, block ports for all others, |
| 16:30.04 | Kobaz | yeah |
| 16:30.08 | Kobaz | allow-only on specific ips |
| 16:33.40 | navaismo | So are we deployin Opus in Asterisk without patches already or I need to wait another 2 years?? |
| 16:34.00 | *** join/#asterisk fornax (~fornax@85.183.43.221) |
| 16:35.20 | *** join/#asterisk Get_The_Fish (~get_the_f@c-24-9-79-75.hsd1.co.comcast.net) |
| 16:35.44 | Get_The_Fish | How can I go about getting the DPMA module compiled for ARM7? |
| 16:36.23 | WIMPy | I guess you mean ARMv7? |
| 16:36.41 | Get_The_Fish | Sorry yeah |
| 16:37.37 | WIMPy | Not much luck, I fear. |
| 16:38.15 | Get_The_Fish | WIMPy: May |
| 16:38.31 | Get_The_Fish | WIMPy: May I ask why that is? Is there a technical reason there? |
| 16:39.04 | WIMPy | Don't think so, but someone has to find the time to do it. |
| 16:40.13 | Get_The_Fish | WIMPy: Gotcha. So it's probably just a matter of compiling it on that processor, correct? |
| 16:40.37 | WIMPy | Hopefully, yes. |
| 16:41.04 | WIMPy | The rest of Asterisk seems to play nicely. |
| 16:41.46 | drmessano | Get_The_Fish, might be worth a phone call to Digium support |
| 16:42.12 | drmessano | You bought the phones.. the phones are supported.. It's a legit request to ask them to queue up compiling it on ARMv7 |
| 16:42.18 | drmessano | Whether or not they do it.. IDK |
| 16:43.17 | Get_The_Fish | drmessano: Thanks, I'll try emailing support first. I wouldnt think that it would be that difficult of a request to accomodate. |
| 16:43.23 | WIMPy | I'd prefer to be able to build dahdo on ARM. |
| 16:43.42 | WIMPy | dahdi, off course. |
| 16:44.00 | Get_The_Fish | WIMPy: I was thinking about asking about that one too. |
| 16:44.14 | drmessano | Youre better off calling, IMO |
| 16:44.18 | drmessano | Nobody reads email anymor |
| 16:44.20 | drmessano | Nobody reads email anymore |
| 16:44.29 | drmessano | You're just as likely to DM them on twitter |
| 16:45.04 | drmessano | E-mail is the new Fax |
| 16:45.28 | WIMPy | Oh, so that's the reason, people want fax? |
| 16:46.21 | navaismo | WIMPy: You can compile dahdi on arm |
| 16:46.30 | drmessano | OMG it's navaismo |
| 16:46.34 | navaismo | hehe |
| 16:46.36 | drmessano | I say "twitter" and boom |
| 16:46.36 | navaismo | at least on the RPi |
| 16:46.43 | navaismo | HII o/ |
| 16:46.49 | WIMPy | At least on raspbian it failed miserably. |
| 16:46.50 | navaismo | hahaha |
| 16:46.52 | drmessano | Howdy fellow twitterer |
| 16:47.11 | WIMPy | Hmm. |
| 16:47.21 | WIMPy | What did you do to make it work? |
| 16:47.23 | Get_The_Fish | Interesting |
| 16:47.29 | navaismo | WIMPy: In fedora I nedded the kernel sources and much patience |
| 16:47.58 | drmessano | DAHDI on an RPi is interesting.. Since you can't put a TDM card in a RPi, do you shove it in the smartjack enclosure and keep hitting it with a hammer til the door closes? |
| 16:48.09 | WIMPy | They didn't get me much further :-( |
| 16:48.23 | WIMPy | No, but you could connect an Astribank. |
| 16:48.25 | navaismo | I tried the sangoma usbfxo |
| 16:48.34 | drmessano | True true |
| 16:48.47 | navaismo | but NEIN |
| 16:49.06 | navaismo | WIMPy: I made this https://www.raspberrypi.org/forums/viewtopic.php?t=17322&p=509680 |
| 16:49.13 | drmessano | I now have a Rhino PRI channelbank to play with but lack another PRI card |
| 16:49.20 | Get_The_Fish | I'm betting it's due to the lack of a real time clock on the RPi |
| 16:49.25 | navaismo | 3 years ago OMG i dont feel so old |
| 16:50.10 | drmessano | hahahah |
| 16:50.17 | drmessano | I was JOKING, but this is funny |
| 16:50.18 | WIMPy | Realtime clock shouldn't matter. You jsut need a ticker and that shouldn't be worse than elsewhere. |
| 16:50.19 | drmessano | by tgwaste » Thu Jan 31, 2013 7:17 pm |
| 16:50.20 | drmessano | Hi, |
| 16:50.20 | drmessano | I dont understand. How does the raspberry pi board receive calls? Connecting a digium card to it somehow? |
| 16:50.54 | drmessano | In the words of [TK]D-Fender "Lots of solder" |
| 16:51.08 | navaismo | haha |
| 16:51.22 | navaismo | i have an astribank hmmm maybe its time to try again |
| 16:51.47 | WIMPy | So so I. That's why I tried dahdi. |
| 16:52.13 | drmessano | That would be a fun project |
| 16:52.22 | navaismo | i did it for the sangoma usbfxo |
| 16:52.25 | navaismo | but stupid wanpipe |
| 16:52.33 | navaismo | hates me |
| 16:53.23 | WIMPy | USB BRI was easy. So I already have a hardware interface on it, but you can always take it a step further :-) |
| 16:54.54 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:55.37 | navaismo | https://pbs.twimg.com/media/Bj_4yGZCAAAsDM-.png:large |
| 16:56.20 | navaismo | for things like that I decided to make&sell cakes instead technology |
| 16:58.24 | *** part/#asterisk monsterco (~monsterco@70.49.229.188) |
| 16:58.28 | WIMPy | Yeah, I'm glad I never got hold of a Sangoma card. I find dahdi hard enough, adding wanpipe on top doesn't sound like anythign I want to get in to. |
| 16:58.30 | *** join/#asterisk Encrypt (~Chuck@cpc65608-newt33-2-0-cust287.19-3.cable.virginm.net) |
| 16:58.32 | Encrypt | Hi again |
| 16:58.43 | Encrypt | Sorry, my internet connection is just horrible |
| 16:59.05 | Encrypt | We have an old Netgear router here and the WiFi randomly drops... |
| 16:59.07 | Encrypt | (T.T) |
| 16:59.11 | WIMPy | (Even if their hardware used to be the best) |
| 16:59.28 | Encrypt | Actually, that's because of a faulty firmware version |
| 16:59.54 | Encrypt | But since we are renting a house with Wi-Fi, we can't flash the firmware because Virgin wouldn't like it... |
| 17:00.04 | Encrypt | And so would the landlord |
| 17:01.04 | navaismo | don't ask Just do it and when you finish you only move on |
| 17:01.11 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:01.49 | navaismo | they will be gratefull forever with you. There is nothing more disgusting like a faulty WiFi |
| 17:03.49 | WIMPy | I get a Syntax error from genksyms. |
| 17:05.20 | Encrypt | navaismo, :p |
| 17:06.49 | *** join/#asterisk fornax (~fornax@85.183.43.221) |
| 17:06.58 | Encrypt | So |
| 17:07.08 | Encrypt | What would you do in my case? |
| 17:07.17 | Encrypt | In order to connect to my remote asterisk? |
| 17:07.48 | Encrypt | Open the appropriate ports on the router (SIP + RTP) and set iptable rules so that I am the only one able to connect? |
| 17:07.59 | *** join/#asterisk WangDang (~wangdang@24.140.224.98) |
| 17:08.08 | navaismo | I choose the vpn option |
| 17:08.29 | Encrypt | Ok |
| 17:08.35 | navaismo | using openvpn and a windows/linux client or yealink phone |
| 17:08.49 | Encrypt | navaismo, Both are running Linux |
| 17:09.01 | Encrypt | More precisele Debian |
| 17:09.05 | Encrypt | precisely* |
| 17:09.13 | WIMPy | Gesundheit! |
| 17:09.14 | Encrypt | So, openvpn is the best solution? |
| 17:09.28 | Encrypt | Do you have any doc which would help me to do it? |
| 17:10.49 | WIMPy | Anyone know how I can make make verbose on dahdi-linux? |
| 17:10.53 | navaismo | not in hand but google is the god of the answers |
| 17:10.59 | navaismo | so go ahead |
| 17:10.59 | Encrypt | Yep |
| 17:11.05 | Encrypt | I managed to find som |
| 17:11.06 | Encrypt | +e |
| 17:11.11 | Encrypt | Thanks |
| 17:11.37 | WIMPy | "make" is a bad term to google for. |
| 17:12.11 | navaismo | I should talk with my exboss to realease the elastix openvpn for all platforms. |
| 17:12.45 | navaismo | s/openvpn/openvpn addon/ |
| 17:22.18 | *** join/#asterisk averythomas (~averythom@2607:5300:60:2d42::1) |
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| 17:44.10 | *** join/#asterisk emdk (~emdk@50-78-2-141-static.hfc.comcastbusiness.net) |
| 17:44.59 | emdk | Hi |
| 17:45.33 | emdk | Is there better support for H.323 than the chan_ooh323 channel driver? |
| 17:48.50 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:50.23 | nunopereira | we have sounds and recordings saved on a mounted nfs directory |
| 17:50.52 | *** join/#asterisk fergus (~sergei@178.120.200.146) |
| 17:51.35 | nunopereira | is that a problem for asterisk to know if those dirs have been updated? |
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| 18:14.42 | jpsharp | nunopereira: As long as the time on your NFS server matches your Asterisk box, you should be fine. |
| 18:14.52 | jpsharp | NTP is your friend. |
| 18:16.30 | *** part/#asterisk killermoehre (~killermoe@lostbit.de) |
| 18:20.25 | nunopereira | @jsharp: so, that's strange that I have get "File not found" for uploaded files |
| 18:20.26 | jpsharp | And I need simple "yes (or 'yeah','sure','ok') / no" speech recognition for Asterisk. Suggestions? |
| 18:20.50 | nunopereira | but after a restart of asterisk it gets back to work |
| 18:22.10 | [TK]D-Fender | jpsharp: sphinx |
| 18:23.02 | jpsharp | I was afraid someone would say sphinx :) |
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| 19:25.30 | shudon | hi all :) book recommendations for newb? |
| 19:25.47 | WIMPy | ~book |
| 19:25.47 | 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 |
| 19:25.52 | shudon | WIMPy: thanks :) |
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| 19:49.14 | mutn | hey guys, using the AGI, what command do I do if I want to stream a file but also accept DTMF? non-blocking you could say |
| 19:49.31 | mutn | right now if I stream file it has to finish before the next line in my script is executed (check digit) |
| 19:49.54 | WIMPy | navaismo: Looks better after I did some upgrading and changed to dahdi from git, |
| 19:50.51 | [TK]D-Fender | mutn: stream file allows you to escape on digit and returns it as well... |
| 19:50.55 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/AGICommand_stream+file |
| 19:56.08 | WIMPy | Nope modpost failed :-( |
| 20:05.24 | *** join/#asterisk Nephfl (6b9005ae@gateway/web/freenode/ip.107.144.5.174) |
| 20:06.59 | Nephfl | Hello, I wrote an agi that uses the originate command to connect an external person/number to a confbridge application. When I run this, it functions as I would expect, but is there a command to set to return them to the dialplan so that if they hang up, they will hit the h extension? |
| 20:09.05 | [TK]D-Fender | Nephfl: It is a channel with all the rules of any other channel. |
| 20:09.11 | [TK]D-Fender | "h" applies. |
| 20:09.33 | [TK]D-Fender | So do other dialplan app options related to this |
| 20:09.46 | Nephfl | it never executes the h from the context set in in the originate command |
| 20:11.12 | Nephfl | I set a system call to a code that outputs to a log in the h extension, an normal transfer to the context and hangup calls the h and it outputs to the log, when I do it with the originate command, it just shows that it is playing the exit message in confbridge but never calls my h extension |
| 20:13.02 | Nephfl | exten => h,1, system() call to a php code that uses error_log() to write to a file |
| 20:15.44 | [TK]D-Fender | H is always executed in the context the call was IN. Not where it STARTED |
| 20:15.45 | Nephfl | and when I am passed into the context from an extension in the system, and hang up, I get the normal h:system() extecuted...but from originate using php-asmanager with channel local/<outbound>@<mycontext>,<exten>,<mycontext>,... ,confbridge,<myconf> and then hang up, nothing shows in cli and my h app isn't run |
| 20:16.13 | [TK]D-Fender | You are also specifying both a context AND an application there. |
| 20:16.15 | [TK]D-Fender | NOT GOOD |
| 20:16.43 | Nephfl | so, what context would it connect to with null values? |
| 20:17.07 | Nephfl | the confbridge,<context> one? |
| 20:18.17 | Nephfl | ok..I have an idea to try |
| 20:18.17 | [TK]D-Fender | why are we seeing confbridge at all? |
| 20:18.27 | [TK]D-Fender | You should be dumping them in the DIALPLAN.... |
| 20:18.44 | Nephfl | what do you mean? |
| 20:19.16 | Nephfl | you mean set <context>,<exten>,<priority> and leave <app> null ? |
| 20:19.25 | [TK]D-Fender | YES |
| 20:20.15 | Nephfl | I'll give it a try... I obviously don't play with agi much, so am still figuring out the basic comands |
| 20:21.01 | Nephfl | you have given me several approaches to check |
| 20:21.04 | Nephfl | thanks |
| 20:21.23 | [TK]D-Fender | That's AMI, not AGI |
| 20:22.15 | WIMPy | If it's AMI, using a System call from dialplan looks rather silly. |
| 20:24.01 | [TK]D-Fender | AMI Originate to launch a channel, but the System call is to log something once the call is done |
| 20:24.04 | [TK]D-Fender | perfectly normal |
| 20:24.07 | Nephfl | it is ami |
| 20:24.12 | Nephfl | well, there is agi and ami |
| 20:24.44 | [TK]D-Fender | checkout time, BBIABn |
| 20:25.01 | Nephfl | what I need to do on hangup is move some files around, so i thought maybe to do a system(script &) to keep it from holding up |
| 20:25.16 | Nephfl | hmm |
| 20:25.46 | WIMPy | Why do people alwyas think AMI is write-only? |
| 20:26.04 | Nephfl | that isn't why I would use ami |
| 20:26.28 | Nephfl | I would use ami because I need to check the status of multiple channels and don't need to handle the current call..I just need the current call to trigger a script |
| 20:26.39 | Nephfl | I'm not doing an interactive agi type script |
| 20:27.11 | Nephfl | it is a call that kicks off dialing multiple parties and connects them to confbridge, then does some work on hangup of all parties |
| 20:27.27 | WIMPy | AMI is not usable for what AGI does. |
| 20:27.28 | Nephfl | I could do it all in dialplan but I'm working on existing stuff |
| 20:28.17 | Nephfl | hmm.... specifically, I am using php-asmanager I haven't looked much into its method for connection |
| 20:29.20 | WIMPy | AGI works on an existing call. AMI allows you to manipulate many things an it will tell you tons about what's going on. |
| 20:29.42 | WIMPy | So rather different stuff. |
| 20:30.06 | Nephfl | looks like it is using AGI |
| 20:31.31 | Nephfl | looks like phpagi-asmanager provides some methods that work on the AGI object...to do things like set handlers for events and provide a method for originate rather than doing a straight string connection |
| 20:34.02 | Nephfl | lol, looks like that fixed my issue, now I need to figure out why I'm dialing twice |
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| 21:12.19 | litn | I'm trying to pass this call off to another extension in an AGI script, it appears I have the wrong command? |
| 21:12.22 | litn | AGI Tx >> 510 Invalid or unknown command |
| 21:12.25 | litn | AGI Rx << DIAL SIP/951 |
| 21:16.35 | navaismo | ~pastebin |
| 21:16.35 | 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. |
| 21:21.59 | [TK]D-Fender | It isn't a valid command. |
| 21:22.10 | [TK]D-Fender | Go read the command list |
| 21:22.31 | litn | thanks, I got it now |
| 21:22.33 | litn | had to do "exec dial" |
| 22:01.08 | *** join/#asterisk moe` (~moe@node-be028882.scarlet.cw) |
| 22:01.14 | moe` | hey guys |
| 22:01.29 | moe` | so not being too drunk, I figured out voicemail. |
| 22:01.32 | moe` | that's a nice thing. |
| 22:02.16 | moe` | now a question... if you call your own number/extension, while its ringing is it possible that the user presses # to get to the voicemail menu (with PIN requirement of course) |
| 22:03.00 | moe` | right now I have it setup so a public number rings various devices for 30 seconds |
| 22:03.05 | moe` | then it goes to voicemail |
| 22:03.06 | moe` | that works |
| 22:03.44 | moe` | the to check the voicemail, the user has to dial another extension... both internal extension (4 digit) and an actual landline that goes to VoiceMailMain() |
| 22:04.27 | moe` | it would be a nice thing if the user could simply dial their own extension (4 digit asterisk context or landline via SIP trunk) and press # to get to VoiceMailMain() |
| 22:05.16 | moe` | any suggestions? |
| 22:05.26 | moe` | I'm sure its something stupid I've overlooked. |
| 22:05.45 | WIMPy | Yes. 'core show application voicemail'. Check the options. |
| 22:06.56 | moe` | uh, that doesn't help me (in my stupidity) |
| 22:07.26 | moe` | I see its likely an options item for the VoiceMail statement |
| 22:07.33 | moe` | ? |
| 22:07.49 | WIMPy | That's what I suggested looking at. |
| 22:08.30 | moe` | ok, so baby step me |
| 22:08.42 | WIMPy | Not a parameter. But there are two special extensions availabe. |
| 22:09.34 | moe` | well I want it so that the user calls their own number (internal extension or public number) and can # drop into their voicemail. |
| 22:10.15 | WIMPy | You only have 0 and *. |
| 22:10.22 | moe` | ok |
| 22:10.23 | [TK]D-Fender | Read up on the4 list of Asterisk Standard Extensions |
| 22:10.33 | WIMPy | Or you build somethign of your own in the dialplan. |
| 22:11.18 | WIMPy | Now I managed to install dahdi modules that cannot be loaded... |
| 22:11.30 | WIMPy | But I'm getting closer. |
| 22:11.40 | WIMPy | (or so I hope) |
| 22:14.08 | moe` | how does one define an exten so it calls another function when the user presses 0 or * while ringing? |
| 22:14.12 | moe` | that's what I'm after |
| 22:14.44 | [TK]D-Fender | you make it |
| 22:14.54 | [TK]D-Fender | And you read the voicemail.conf.sample |
| 22:14.56 | moe` | and how would one do that [TK]D-Fender ? |
| 22:14.59 | WIMPy | You don't. You wait until you get in to VoiceMail. |
| 22:15.04 | [TK]D-Fender | <[TK]D-Fender> Read up on the4 list of Asterisk Standard Extensions <- |
| 22:15.24 | [TK]D-Fender | While RINGING would require that DIAL supports it |
| 22:15.29 | [TK]D-Fender | READ DIAL'S INSTRUCTIONS |
| 22:16.01 | moe` | yeah I hear what you're saying |
| 22:16.11 | WIMPy | And it also requires answering the call before dialling. |
| 22:16.19 | moe` | but being a little bit too much into the beers, I'm wondering of an exten syntax example |
| 22:16.31 | [TK]D-Fender | "Instructions HAVE samples |
| 22:16.33 | [TK]D-Fender | READ THEM |
| 22:16.36 | [TK]D-Fender | No more excuses |
| 22:16.42 | [TK]D-Fender | "core show application dial" <- |
| 22:18.29 | *** part/#asterisk kharwell (kharwell@nat/digium/x-fekldqtxjoogiwpf) |
| 22:19.32 | moe` | ok, so a "d" option on the exten |
| 22:19.36 | moe` | working on it... |
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| 22:31.45 | WIMPy | is giving up for today. Maybe I should try to cross compile the dahdi modules? |
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| 23:03.20 | moe` | still haven't found the magic to have the user press 0 or * during a ring to sling off to a different function |
| 23:03.27 | moe` | exten examples anyone? |
| 23:04.09 | moe` | its probably something really simple that my OCD overlooks :) |
| 23:06.41 | [TK]D-Fender | If you were OCD you'd fave read the instructions 20 times |
| 23:06.56 | [TK]D-Fender | d: Allow the calling user to dial a 1 digit extension while waiting |
| 23:06.56 | [TK]D-Fender | <PROTECTED> |
| 23:06.56 | [TK]D-Fender | <PROTECTED> |
| 23:06.56 | [TK]D-Fender | <PROTECTED> |
| 23:07.04 | [TK]D-Fender | Wha tis NOT CLEAR about this? |
| 23:07.25 | [TK]D-Fender | Do you HAVE an exten to match the digit dialed in the PROPER context? |
| 23:08.21 | [TK]D-Fender | "Exit to that extension if it exists in the current context, or the context defined in the ${EXITCONTEXT} variable, if it exists." <- this is EXTREMEMLY clear about what you need to have to match it. |
| 23:09.01 | [TK]D-Fender | an exten in the CURRENT context, or in the one that ${EXITCONTEXT} points to. |
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| 23:21.53 | moe` | [TK]D-Fender, sorry man, I'm just useless at the moment. I see what has to be done. |
| 23:23.22 | moe` | I know it's a "d" option, so perhaps tomorrow I'll enable that. If you recall a while ago I was being stupid about voicemail, but after 5 minutes I sorted it when I had a mind to focus. |
| 23:23.51 | moe` | its actually really cool how asterisk handles voicemail |
| 23:25.24 | moe` | the config file parsing code for asterisk must be awesome... |
| 23:26.16 | moe` | are there any asterisk developers here? |
| 23:26.49 | moe` | if so, I'll give you a thumbs up... good software. |
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