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| 03:15.10 | winadminguy | I just installed asterisk and bought DID's and outbound credits with flowroute. When I call my self, the call connects I see data in debug mode but it auto hangs up on me. Can anyone assist me? |
| 03:16.05 | *** join/#asterisk Gallomimia (~quassel@S010684948cbeac73.ca.shawcable.net) |
| 03:21.14 | winadminguy | exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@flowroute) |
| 03:21.28 | winadminguy | do I replace any values in that? I got that config from flowroute |
| 03:21.51 | winadminguy | do I replace the _1(my cell number here) ? |
| 03:21.56 | Samot | You have to write the dialplan. |
| 03:21.59 | Samot | No. |
| 03:22.13 | Samot | http://wiki.asterisk.org <-- You need to start there. |
| 03:22.39 | Samot | Asterisk is 100% on you to setup the configuration files with your specific information. |
| 03:23.08 | Samot | ${EXTEN} is a system variable. |
| 03:23.40 | Samot | Whatever 1NXXNXXXXXX is, that's the value of ${EXTEN} |
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| 03:40.21 | Gallomimia | so uh... i started reading a pdf book i got online. then i realized it was 10 years old... should i start over with an up to date version? |
| 03:41.58 | Samot | ~book |
| 03:41.58 | 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 |
| 03:42.16 | winadminguy | I've got it registered, how can I configure the dial plan to ring my cell phone? |
| 03:42.26 | Samot | Most of the dialplan stuff hasn't changed. |
| 03:42.41 | Samot | Between the wiki, sample configs and that you should be fine. |
| 03:43.05 | Samot | winadminguy: That's not how it works. |
| 03:43.22 | Samot | I've provided three prime sources for learning/using Asterisk. |
| 03:43.40 | winadminguy | what do you mean samot? |
| 03:43.52 | Samot | "How do I do X?" |
| 03:43.52 | winadminguy | when I get an incoming call I want it to ring my cellphone |
| 03:43.59 | Samot | Followed by "How do I do Y?" |
| 03:44.06 | Samot | Followed by "How do I do Z?" |
| 03:44.13 | Samot | It's not hand holding step by step |
| 03:44.19 | Samot | Wiki, Book, Sample configs. |
| 03:44.28 | winadminguy | ok, how do I configure asterisk to allow me to call in and set a voice greeting for the pbx |
| 03:44.31 | Samot | All three of those will answer all the basics |
| 03:44.41 | Samot | Go read. |
| 03:44.43 | Samot | Learn |
| 03:44.48 | Samot | Try |
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| 09:06.16 | x86- | can you direct me to documentation on how to set a pin number when a certain extension is reached, if the pin is incorrect it disconnects if the pin is correct it proceeds? |
| 09:19.25 | tuxd00d | https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_Authenticate |
| 09:19.32 | tuxd00d | https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_VMAuthenticate |
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| 09:28.23 | tuxd00d | x86-: Three failed attemps will hangup |
| 09:36.48 | x86- | so for example Authenticate(12345,[3,[prompt]]]) ? |
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| 10:03.33 | tuxd00d | same => n,Authenticate(1234) |
| 10:05.06 | tuxd00d | or same => n,Authenticate(1234,,4) ; if you want to make the # optional |
| 10:08.40 | tuxd00d | It automatically plays a âplease enter your password, followed by the pound keyâ. If you want to change the instructions, you specify it |
| 10:08.59 | tuxd00d | same => n,Authenticate(1234,,,auth_instructions) |
| 10:16.35 | x86- | thanks tuxd00d :) |
| 10:16.55 | tuxd00d | :) |
| 10:17.38 | x86- | can you direct me to more scripting examples |
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| 10:22.25 | tuxd00d | ~book |
| 10:22.25 | 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 |
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| 14:04.16 | Samot | Lol. |
| 14:20.37 | Samot | tuxd00d: You gots played. |
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| 14:40.28 | Ox21O2 | Hey. What is a good SIP account provider I can use with asterisk to do international calls etc? |
| 14:40.54 | fauxalliance | flowroute, voip.ms, didlogic |
| 14:41.22 | fauxalliance | shop around |
| 14:41.38 | Ox21O2 | thanks |
| 14:54.40 | voipmonk | Where are your top 10 destinations? |
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| 15:24.03 | youtmon | good day |
| 15:25.45 | youtmon | having is issue I'm unable to make calls, getting busy and seeing log ,message "Unable to retrieve PJSIP transport 'udp,tcp,ws,wss' for endpoint 'anonymous", what could this be? |
| 15:31.19 | file | you are trying to use the "transport" option like chan_sip and it doesn't work like that in chan_pjsip |
| 15:31.24 | file | remove it and it'll work |
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| 15:57.36 | youtmon | am using PJSIP on the extension, and transport is set to auto, |
| 15:58.06 | youtmon | still getting the error |
| 15:58.14 | youtmon | this started out of the blue |
| 16:00.56 | file | those values are from chan_sip... they won't work in chan_pjsip, so I don't really know what to say... something must have changed |
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| 17:34.52 | Samot | Well, do you have an endpoint named 'anonymous'? |
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| 20:54.31 | _Maik_ | hello, i have a problem using asterisk with yealink sip phones. i get Failed to authenticate on INVITE to '"phone01" <sip:620@fritz.box>;tag=as764f90a8' |
| 20:58.23 | [TK]D-Fender | Fritz != yealink |
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| 20:59.33 | _Maik_ | yea but the "phone01" drops the error |
| 20:59.46 | [TK]D-Fender | Show the actual full call attempt w/ sip debug |
| 21:01.09 | [TK]D-Fender | !pb |
| 21:01.13 | [TK]D-Fender | ~pb |
| 21:01.13 | infobot | extra, extra, read all about it, pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 21:01.14 | [TK]D-Fender | ^^^ |
| 21:03.09 | _Maik_ | https://pastebin.com/uxXesS43 |
| 21:07.55 | _Maik_ | sip.conf https://pastebin.com/kjVP0JMJ |
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| 21:13.53 | _Maik_ | [TK]D-Fender: do you see any config failures? |
| 21:15.02 | Samot | Everything is incomplete. |
| 21:15.16 | Samot | Those configs are incomplete |
| 21:15.27 | Samot | The debug for the INVITE isn't full.. |
| 21:16.02 | _Maik_ | thats all what asterisk gives me @ sip debug |
| 21:16.50 | [TK]D-Fender | "sip show peer phone01" <- |
| 21:18.35 | _Maik_ | https://pastebin.com/A80sS0H2 |
| 21:18.59 | [TK]D-Fender | "sip show peers |
| 21:19.48 | Samot | There should have been a second INVITE after that 401 reply |
| 21:20.01 | _Maik_ | https://pastebin.com/qxMLZhgc |
| 21:20.02 | Samot | With the actually user/pass digest as part of it |
| 21:20.48 | Samot | allow=g722,ulaw,alaw <-- This isn't going to work without disallow=all above it |
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| 21:21.13 | Samot | Your phones are not behind NAT so you should have nat=no in those. |
| 21:22.24 | _Maik_ | Samot: https://pastebin.com/CJFSQykM |
| 21:22.29 | Samot | directmedia=no should be with the phone configs. |
| 21:22.34 | Samot | Because by default it is "yes" |
| 21:23.32 | Samot | realm="fritz.box" <-- I hate those things. |
| 21:23.47 | Samot | So your FritzBox is totally sending the wrong auth realm Asterisk expects. |
| 21:23.53 | Samot | So yeah, never going to auth that call. |
| 21:24.09 | _Maik_ | incomming works ^^ |
| 21:24.17 | Samot | Incoming isn't authed |
| 21:24.27 | _Maik_ | yea |
| 21:24.40 | _Maik_ | how to get outgoing working? |
| 21:24.40 | Samot | Incoming to your phone is Asterisk ->> Phone |
| 21:25.02 | Samot | Well, that's a FritzBox issue |
| 21:25.33 | _Maik_ | no |
| 21:26.05 | _Maik_ | i have a other asterisk server running with 2 phones working in&out bound |
| 21:26.32 | _Maik_ | same version |
| 21:26.44 | Samot | OK then compare the boxes. |
| 21:27.15 | _Maik_ | Asterisk 13.17.1 |
| 21:27.23 | Samot | The configs. |
| 21:27.24 | Samot | The calls |
| 21:27.32 | Samot | You're going to have to see what is different. |
| 21:27.35 | Samot | We can't tell you that. |
| 21:27.45 | _Maik_ | the difference ist the phone i am using |
| 21:28.02 | Samot | Well... |
| 21:28.09 | Samot | did you use the exact same config from the other box? |
| 21:28.24 | _Maik_ | yes |
| 21:28.39 | _Maik_ | only change was user/password |
| 21:30.49 | Samot | Well, can't be much help myself. My only experience with FritzBoxes is people in here trying to make them work properly with Asterisk. |
| 21:31.23 | Samot | So if your theory is it's the phone, then use that phone on the working box. |
| 21:31.29 | Samot | If it works, then it's not the phone. |
| 21:32.05 | Samot | Then taking a phone from the working box and put it on the none working box, if it doesn't work then it's the box |
| 21:32.11 | Samot | Or the config of the new setup |
| 21:34.35 | _Maik_ | thats all i checked |
| 21:34.45 | _Maik_ | cant find the iseue |
| 21:34.53 | Samot | So you've done both those things? |
| 21:35.05 | Samot | Put this phone you claim is the issue on the working box and it works? |
| 21:35.29 | Samot | The took a phone from the working box, put it on the this one and you could make outbound calls? |
| 21:39.42 | _Maik_ | cant connect the phone to the working box, is in an other office and vlan protected |
| 21:43.11 | voipmonk | Telecom and networking go together like law and history. Get to know your networking team. Can't isn't going to help you. |
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| 21:44.25 | _Maik_ | @voipmonk: ask vodafone....there yuo get a no |
| 21:45.22 | voipmonk | So many other options I'm out sure why you brought that up. |
| 21:46.20 | voipmonk | s/out/not |
| 21:47.07 | _Maik_ | i am out of ideas what can fix the isue |
| 21:48.07 | _Maik_ | maybe so "he did you tried this" ... |
| 21:50.24 | Samot | Well the issue is between the friztbox and the PBX |
| 21:50.29 | Samot | Like I said. |
| 21:50.43 | Samot | Since the Friztbox is what is denying the outbound call from the PBX |
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