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| 00:19.19 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.2 (2018/02/21), Standard: 15.2.2 (2018/02/21); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 09:08.20 | Sepultura | file, is the new Raspberry Pi strong Enough for Asterisk for a 5-10 Participant phone conference? |
| 09:14.50 | Samot | What does it have |
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| 10:57.20 | Ian_ | Faster ethernet it would seem. Released yesterday from all accounts. Thanks for bringing this new baby to my attention. Ordering time for Ian_; add to the cotery |
| 10:57.26 | Ian_ | 14 March 2018 - Raspberry Pi Model 3 B+ announced with faster CPU speed, faster ethernet, and updated WiFi and Bluetooth capabilities |
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| 11:18.36 | sibiria | faster ethernet - but still attached on USB2, and possibly still with the old USB2 rate saturation problems |
| 11:18.44 | sibiria | also still 1GB RAM |
| 11:19.31 | sibiria | unless you absolutely need GPU hardware acceleration on Linux, or absolutely must have a Pi, there are better boards for same and less money |
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| 11:20.26 | sibiria | Sepultura: yes it's fast enough for 10 party conf - and more |
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| 11:21.07 | sibiria | the low 1GB RAM is your main problem, depending on what features you're gonna be running through and on Asterisk |
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| 11:22.40 | Sepultura | sibiria thx :) |
| 11:22.47 | Sepultura | sibiria are you from Mother Russia? |
| 11:22.50 | Sepultura | :) |
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| 11:23.36 | Sepultura | sibiria which board is better? Banana Pi? |
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| 11:24.04 | sibiria | i am not from russia, no :) |
| 11:24.18 | sibiria | i have the Pine64, and the ODroid C2 |
| 11:24.25 | sibiria | the latter is a bit more expensive today, but worth the money |
| 11:26.14 | Sepultura | Odroid C2 sounds good |
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| 11:26.49 | sibiria | the 2GB of RAM on it makes a world of difference, enables so much more practical use |
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| 11:28.11 | sibiria | same regarding the eMMC port |
| 11:28.26 | sibiria | huge difference in i/o speed |
| 11:28.34 | Sepultura | sibiria but if you are not from Russia your why did you choose the nick? |
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| 11:29.01 | Samot | Gigabit Ethernet with 300Mbps throughput. |
| 11:29.02 | Samot | Lol |
| 11:29.12 | sibiria | Sepultura: because i like how it rolls off my tongue |
| 11:30.04 | sibiria | Samot: it's a bit ironic :) |
| 11:30.19 | sibiria | but to the defense of USB2-based "gigabit" ethernet, 300mbit/s goes a long way |
| 11:30.26 | sibiria | more so than "just" 100mbit |
| 11:30.44 | sibiria | though i wonder if the Pi 3 b+ really can saturate those 300mbit/s |
| 11:31.03 | sibiria | the previous Pi models had massive problems saturing the USB2 interface |
| 11:31.06 | sibiria | saturating* |
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| 11:40.15 | Sepultura | sibiria I need to build a Asterisk Phonebox for Telephone Conference with 10 Persions :P |
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| 12:06.50 | sibiria | Sepultura: even an older Raspberry Pi would work, but 1GB really is the limit of what you want to run Linux + some service on |
| 12:07.16 | Sepultura | I will get a single-board with 2 GB :P |
| 12:07.17 | sibiria | i have no problems running 25 lines on asterisk on my Pine64. no CPU load to mention |
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| 12:26.33 | ruied | Hi. In .call files I want to dial to two extensions simultaneously. Is it possible? I've tried Channel: SIP/310&IAX2/310 with no success |
| 12:27.50 | sibiria | my memory could be off, but with chan_sip isn't that done using hyphens? |
| 12:28.56 | file | no, call files do not directly allow dialing multiple at once |
| 12:29.04 | file | you have to use a Local channel which executes dialplan to do it |
| 12:29.49 | ruied | ok |
| 12:30.00 | ruied | thanks |
| 12:30.43 | sibiria | ah, maybe i was thinking about using chan_sip to pick a random trunk |
| 12:31.17 | [TK]D-Fender | chan_sip doesn't make decisions |
| 12:32.04 | sibiria | i'm pretty sure we used some sort of simplistic "load balancer" in 1.8 |
| 12:32.22 | sibiria | by specifying a set of trunks for the Channel variable in the call-file |
| 12:32.58 | sibiria | making chan_sip on its own pick on random in the set |
| 12:33.45 | Samot | How does Chan_SIP pick it? |
| 12:34.01 | Samot | Chan_SIP isn't called until you call with with a Dial(). |
| 12:35.24 | [TK]D-Fender | Chan_sip doesn't use variables to decide. Whatever specifies a Dial passes a complete device dial string |
| 12:36.17 | Samot | Well "load balancer" sounds like the use of groups and checking how many calls in in that group. |
| 12:37.32 | sibiria | we didn't run any functionality for that in the dial plan - it was just specifying multiple trunks in the Channel variable of the call file |
| 12:38.33 | Samot | Then I'm not sure how you are doing this. |
| 12:38.38 | [TK]D-Fender | chan_sip isn't doing any deciding though |
| 12:39.11 | Samot | Or the call files |
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| 14:48.23 | LkBurn | good morning ladies and gents |
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| 15:58.23 | comrad | hi, i have a question about the voicemail app option d(c) |
| 15:58.39 | comrad | does this change the number the voicemail app tells "is not available"? |
| 15:58.51 | comrad | https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_VoiceMail |
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| 16:05.26 | [TK]D-Fender | no |
| 16:05.46 | [TK]D-Fender | that has nothing to do with the box # you are directing it to |
| 16:07.00 | comrad | hm |
| 16:07.10 | comrad | i am using pjsip and having cryptic sip identities |
| 16:07.24 | comrad | which makes the announcement pretty useless as it also only voices the numbers |
| 16:12.47 | [TK]D-Fender | Because you didn't record an announcement and tell * to play it |
| 16:12.54 | [TK]D-Fender | this has nothing to do with PJSIP |
| 16:15.11 | comrad | the voicemail-app plays an announcement |
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| 16:15.25 | comrad | but it skips the chars in the endpoint-name |
| 16:15.45 | comrad | like this is my endpoint PJSIP/429ec2bbfb |
| 16:15.52 | comrad | and voicemail says 4292 |
| 16:27.50 | [TK]D-Fender | endpoint has nothing to do with box # |
| 16:28.05 | [TK]D-Fender | I don't know why you ar trying to use those terms interchangably |
| 16:28.10 | [TK]D-Fender | a box number is NUMERIC |
| 16:28.27 | [TK]D-Fender | and has nothing to do with PJSIP or announcements, or anything else at all |
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| 18:08.59 | epl692 | Can an inbound google voice line support fax detection and send a fax to a fax machine? |
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| 19:31.27 | *** topic/#asterisk by bford -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 19:42.07 | jamesaxl | I am about sharing my solution => Asterisk, Perl restful API and postgresql |
| 20:06.25 | igeni | whats a good webbased voip phone |
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| 21:34.54 | Forbidd3n | Hey everyone. I am calling a SIP and passing extra custom variables in the url for example -- sip:5.5.5.5;param1=value1;param2=value2 -- I have an inbound route that handles the request to an AGI(script.php) file. I need to pass param1 and param2 to the PHP file script.php but figure out how to do this. Anyone here that can help me with this, please? |
| 21:35.39 | Forbidd3n | I have tried -- AGI(script.php,${param1},${param2} -- but no luck |
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| 21:42.37 | Micc | Why is it that the adaptive jitter buffer for chan_sip requires chan_iax2? |
| 21:42.53 | Micc | I tried to unload chan_iax and my system crashed. |
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