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| 01:24.42 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.1 (2018/11/14), Security Only: 15.6.2 (2018/11/14); 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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| 10:19.46 | eolz | quick question guys : is it bad to chain iax2 trunks? ie : ipbx-newyork <=iax2=> ipbx-paris <= iax2 => ipbx-london |
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| 13:05.06 | sicelo | between chan_mobile & chan_dongle, which one seems to be most supported and/or forward-looking? |
| 13:05.31 | file | neither, they have no maintainer and noone really touches them |
| 13:06.04 | sicelo | that sucks |
| 13:06.29 | sicelo | anything newer that provides the same functionality maybe? |
| 13:06.42 | file | nope |
| 13:06.47 | file | unless there is something outside of the tree |
| 13:06.52 | stevedavies | @eolz I don't see why you shouldn't do that. But by these days most people are using SIP (PJSIP) |
| 13:07.06 | sibiria | are there any compelling reasons for upgrading from ast 13 to ast 16 besides not being "left behind"? some performance improvements or so that haven't been back-ported to ast 13 |
| 13:08.12 | stevedavies | @sibiria https://wiki.asterisk.org/wiki/display/AST/New+in+16 |
| 13:09.18 | stevedavies | good improvements for video especially with loss on links. PJSIP faster qualifies (oops, chan_sip used to be faster and we can't have that). |
| 13:09.47 | stevedavies | also since 13 better behaviour for back to back SIP (RTP) when there is loss |
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| 13:11.23 | file | disregarding that aspect - you do need to be aware of the support timeline on things, https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 13:12.13 | stevedavies | @file talking to a pretty large sip trunk provider here in SA still rocking 1.4 .... |
| 13:12.25 | file | sure |
| 13:12.59 | file | it's ultimately the choice of everyone, but being informed is important |
| 13:13.09 | sicelo | stevedavies: SA .. mzansi? :) |
| 13:13.20 | stevedavies | obviously i think they should update |
| 13:13.36 | stevedavies | @sicelo yebo |
| 13:14.35 | sicelo | awesome. |
| 13:14.54 | stevedavies | u too? |
| 13:15.27 | sicelo | i'm your neighbor, Eswatini |
| 13:16.43 | stevedavies | aha. Asterisk is everywhere. World domination? Check. |
| 13:18.27 | sicelo | we don't have voip providers here though. i just wanted to use Asterisk privately (buy voice bundles, and share with my family via voip) |
| 13:21.41 | stevedavies | Nice project. Our client Edgars has stores in Eswatini but we don't do voip there. |
| 13:23.57 | sicelo | i used to work for a computer company, and they tried asterisk with a few customers .. didn't seem to work out. not too sure why (i didn't stay in the company for very long) |
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| 20:37.23 | igcewieling | I didn't see anything related to ignoresdpversion in pjsip.conf.sample. does that indicate it is not needed for pjsip? |
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| 21:01.07 | trmg | sicelo: A lot of Asterisk deployments go wrong because the folks who do the implementation don't fully grasp either Asterisk, telephony, or a combination of the two. There are a lot of "barely working" deployments out there because they've been cobbled together and work "good enough", which always makes me sad. |
| 21:01.37 | trmg | Also a good portion of the time it's also because folks are being cheap on how the trunk. |
| 21:01.56 | trmg | Crappy ISPs and no QoS when doing SIP trunking can make the experience poor. |
| 21:02.43 | trmg | With Asterisk, or IP telephony in general, you have to have a pretty solid grasp of networking and telphony...traditionally these roles were separate. |
| 21:05.37 | sicelo | makes sens what you say |
| 21:07.59 | trmg | I have friends in the telephony world who are diehard traditional voice guys (POTS/TDM/ISDN based systems) and swear off VoIP because "it just sucks". This is because they've never experienced a good VoIP implementation. |
| 21:11.03 | csavinovich | @trmg You start feeling that something is gonna go wrong when you are deploying a VoIP system and the client doesn't want to give access to his router. |
| 21:11.19 | trmg | lol |
| 21:11.21 | trmg | Yeah |
| 21:12.02 | csavinovich | I would recommend when you hear that just walk away from the job |
| 21:14.12 | sicelo | i love VoIP and would happily kill off everything else, but then, peoples :) |
| 21:23.27 | trmg | lol |
| 21:23.38 | trmg | I've actually come to appreciate the traditional telephone network. |
| 21:23.46 | trmg | I think they both have their place for sure. |
| 21:28.24 | trmg | I have a little Cisco 2851 with some T1/E1 interfaces and bought a TE210p card for my Asterisk box at home just to play with. |
| 21:28.26 | trmg | It's neat. |
| 21:28.41 | trmg | I drive a bunch of POTS phones using it. |
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