| 00:11.56 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
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| 01:18.53 | Kattyroo | hello my asterisk does not work at all how to fix plz |
| 01:19.57 | WIMPy | Katty: Have you become a bear? |
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| 01:20.23 | joako | I get the warning: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' but I need to match an extension that begins with a character, not a number. |
| 01:20.48 | Penguin | Yeah? So what's the problem? |
| 01:20.49 | WIMPy | [a-zA-Z]? |
| 01:21.09 | Kattyroo | checks |
| 01:21.16 | WIMPy | Well, it is a problem. |
| 01:21.18 | Kattyroo | WIMPy: no, i still appear to be hooman. |
| 01:21.25 | WIMPy | You may want to match more thn just a-z. |
| 01:21.28 | Penguin | Don't use _. because it will match things that you certainly do not want it to match. |
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| 02:20.26 | viktech | hi |
| 02:20.40 | viktech | i am having some basic difficulty in inderstanding |
| 02:21.14 | viktech | where should the asterisk pbx be located |
| 02:21.55 | [TK]D-Fender | Whereve yo find it practical to be located |
| 02:22.03 | [TK]D-Fender | Wherever you find it practical to be located |
| 02:22.59 | viktech | so, let me give examples so i can clarify |
| 02:27.55 | viktech | i can install asterisk on a computer on my network. i can define statics to and from the router and pbx. i can open ports for asterisk services. assuming the configuration of asterisk is sound, is this one way to set up a pbx from which i can call in/out/group ring etc? |
| 02:28.14 | viktech | versus, lets say |
| 02:28.19 | viktech | paid online service |
| 02:28.42 | [TK]D-Fender | What you use it for is up to you. |
| 02:28.49 | [TK]D-Fender | It will work regardless of location. |
| 02:29.06 | viktech | is there a difference between the two |
| 02:29.14 | viktech | like |
| 02:29.22 | viktech | nvm, the answer is no |
| 02:29.26 | viktech | \um |
| 02:29.40 | viktech | can i run asterisk in a vm on my network |
| 02:30.17 | [TK]D-Fender | Yes, but it needs to be tightly tuned to give * highest priority |
| 02:30.30 | [TK]D-Fender | ESXi has some of the best results. |
| 02:30.32 | [TK]D-Fender | YMMV |
| 02:32.09 | viktech | is this implying that a dedicated asterisk server will be better than a virtualized one |
| 02:32.22 | viktech | unless the vm and host are configured correctly for such |
| 02:32.59 | [TK]D-Fender | yes |
| 02:33.14 | [TK]D-Fender | Most spcifically in cases where conferencing is concerned |
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| 02:37.11 | viktech | i see |
| 02:37.16 | viktech | thank you |
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| 05:40.14 | joako | Does anyone how how I can specify a certificate locally instead of putting imapflags = novalidate-cert |
| 05:40.14 | joako | ? |
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| 11:12.59 | liquido_ | Hi, I was wondering if anyone knows whether there's a plan to integrate PPT over Cellular functionality into Asterisk or if there are other possibilities to accomplish PPT over SIP without the use of multicasting? |
| 11:25.01 | SpeedEvil | PPT? |
| 11:25.13 | SpeedEvil | (apologies - I have no idea about the answer, and am just wondering) |
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| 11:55.05 | liquido_ | Push-To-Talk.. similar to a walkie-talkie type functionality |
| 11:55.18 | liquido_ | PTT of course :) |
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| 11:58.36 | SpeedEvil | Ah. |
| 11:59.02 | bluOxigen | hi room |
| 11:59.59 | bluOxigen | got a simple question, whats the best suited sip-phone(hardphone) to use with asterisk, |
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| 12:10.39 | WIMPy | Suitec for what? |
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| 12:22.04 | skirmisha | guys |
| 12:22.29 | skirmisha | what was the option to tell asterisk to auth/match against username first |
| 12:22.53 | skirmisha | i have 3 users from same peer and can't get them working as it match against ip always |
| 12:23.01 | skirmisha | i remember there was an option about this |
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| 12:27.15 | skirmisha | any ideas? |
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| 14:18.26 | eirirs | again nickserv identify, then join, then /window # |
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| 15:53.11 | jkroon | uthorization: Digest username="", realm="polyfonix", nonce="1043f227", uri="sip:192.168.1.3", ... any idea why a polycom would send an empty digest username? |
| 15:55.54 | navaismo | nein |
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| 16:00.08 | pabelanger | because it is empty? |
| 16:00.17 | pabelanger | did you provision using the web interface? |
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| 16:11.14 | qakhan | hi all, is there any way, we can connect 2 calls together. |
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| 16:13.23 | [TK]D-Fender | AMI Bridge |
| 16:15.29 | qakhan | do i have to write script for it? |
| 16:16.22 | [TK]D-Fender | It requires exactly what AMI says it requires. |
| 16:16.33 | [TK]D-Fender | ~book |
| 16:16.33 | 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 |
| 16:16.35 | [TK]D-Fender | ^^^ |
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| 19:51.34 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 19:52.04 | dan_j | Hi, I'm looking at ExtenSpy. The wiki says "Only channels created by outgoing calls for the specified extension will be selected for spying." |
| 19:52.14 | dan_j | How can I set up a spy for an incoming call? |
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| 19:53.44 | [TK]D-Fender | ChanSpy |
| 19:54.44 | dan_j | Ah. Thanks |
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| 20:51.29 | lorsungcu | is there a good way to pause mixmonitor? |
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| 21:01.45 | pabelanger | not that I know off |
| 21:01.48 | pabelanger | aside from stopping it |
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| 21:06.40 | dan_j | When using Chanspy, * seems to hang up the spying channel, instead of moving to the next channel. |
| 21:07.14 | dan_j | The CLI says 'Stopped spying due to the spied-on channel hanging up' but the spied channel isn't hung up. |
| 21:07.37 | dan_j | Its still connected and can be re-spied on by running chanspy again |
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| 21:10.23 | dan_j | It seems that the E option causes a hangup when trying to use '*' to move to the next channel. Bug? |
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| 21:17.28 | SuperNull | if im using info message DTMF .. ultimately how do tones get generated to analog lines ? is the far end digital -> analog expected to generate tones ? (mid-call tones not initial) |
| 21:21.48 | dan_j | Whats the file called that describes all the sounds in asterisk-sounds? |
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| 21:46.24 | leifmadsen | dan_j: sounds.txt? |
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| 21:54.42 | xaristax | I Have turn DTMF debug on asterisk because im having problems with DTMF and i see something strange the dtmf dont work but when i press the digit 2 the call is silence i get this "Failed to initialize UDPTL, declining image stream" |
| 21:54.57 | xaristax | can someone tell me what is this? |
| 21:56.28 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.8.0 (2014/03/03), 1.8.26.0 (2014/03/03); Standard: Asterisk 12.1.0 (2014/03/03); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Please report bullying to an op |
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| 22:06.27 | jmmills | Does any know exactly how asterisk determines which mail sever to use to send a voicemail email? |
| 22:06.55 | jmmills | I've been as of yet unsuccessful to turn on debuging/verbosity for any email subsystem |
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| 22:20.42 | navaismo | ya viste le Lunes de mallitas! |
| 22:39.58 | file | jmmills, it uses the "sendmail" application |
| 22:40.22 | jmmills | true, but is there anything that would let me troubleshoot the system call |
| 22:40.54 | navaismo | maillog |
| 22:40.58 | jmmills | I'm having trouble with a single voicemail box not receiving email and it appears that it's not related to the email system as I've been able to send email to the customer via the sendmail application on the host in question |
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| 22:43.45 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.8.0 (2014/03/03), 1.8.26.0 (2014/03/03); Standard: Asterisk 12.1.0 (2014/03/03); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Please read the Community Code of Conduct at https://wiki.as |
| 22:44.03 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.8.0 (2014/03/03), 1.8.26.0 (2014/03/03); Standard: Asterisk 12.1.0 (2014/03/03); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Please read the Community Code of Conduct at https://wiki.as |
| 22:44.11 | file | mjordan, you have exceed the available space |
| 22:44.14 | file | or exceeded |
| 22:44.16 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.8.0 (2014/03/03), 1.8.26.0 (2014/03/03); Standard: Asterisk 12.1.0 (2014/03/03); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 22:44.19 | mjordan | indeed |
| 22:44.21 | mjordan | oh wellz |
| 22:44.27 | mjordan | ~conduct |
| 22:44.33 | mjordan | k, that's a good one for infobot |
| 22:44.46 | file | jmmills, and not really... you'd have to check your underlying mail transport agent |
| 22:45.04 | file | mjordan, hrm |
| 22:45.09 | jmmills | hmm |
| 22:45.11 | file | mjordan, a bit.ly might fit |
| 22:45.19 | jmmills | file: thank you for the help. |
| 22:45.20 | mjordan | file: true. |
| 22:47.12 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.8.0 (2014/03/03), 1.8.26.0 (2014/03/03); Standard: Asterisk 12.1.0 (2014/03/03); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 23:25.36 | jmetro | yo |
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| 23:28.31 | m0sphere | can anyone help me determine why my asterisk server does not appear to recieve any DTMF tones from any incoming call? My server is set up to accept calls from PSTN via a Custom (not sip) Trunk with Anveo. For testing purposes, the incoming call points to a DISA application with a password. The server receives no response when I enter the password. Here is my CLI http://pastebin.com/w1qu2EdT |
| 23:29.24 | WIMPy | Not SIp, but what? |
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| 23:29.50 | m0sphere | CentOS 6.4 64bit on a VPS with digitalocean. |
| 23:30.30 | m0sphere | when you add a trunk in freepbx, it gives you the option to add a sip, dahdi, iax2, enum, dundi or custom trunk. I have to use custom in order to use Anveo |
| 23:31.53 | WIMPy | If you can't tell us what it is, it's impossible to comment. |
| 23:31.54 | m0sphere | the differences are no peer or user details in the custom trunk selection, and instead of a 'register string', it requires a 'custom dial string' |
| 23:33.56 | WIMPy | If you speak FreePBX, you should ask in #freepbx |
| 23:36.58 | m0sphere | true, i just figured it has something to do with DTMF and thought perhaps somoene in here may be familiar with troubleshooting DTMF issues |
| 23:37.39 | WIMPy | You need to tell us on what techlology/channeltype. |
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