| 00:07.18 | *** join/#asterisk Catatronic (~catatroni@73.93.140.203) |
| 00:09.35 | TandyUK | well voipmonitor is logging |
| 00:09.48 | TandyUK | now to wait for all devices to restart tls sessions so we can actually decrypt them..... |
| 00:10.21 | MrTAP | security's sure a PITA for troubleshooting, eh? |
| 00:10.28 | TandyUK | yeah isnt it just |
| 00:11.57 | Samot | Interesting. |
| 00:12.11 | Samot | I can watch TLS in the console with a SIP debug. |
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| 00:24.44 | hdon | hi all :) i'm setting CDR(dst)=something but it's not being recorded. i can see in my log that the assignment happened, but what's in my database when the call ends does not have this value. http://ix.io/C2D |
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| 00:25.33 | hdon | exten => s,n,Set(CDR(dst)=${OUTNUM}) ; assigning cdr dst field |
| 00:25.50 | hdon | however i'm also setting outbound_cnum and that assignment does "take" |
| 00:28.37 | hdon | or "stick" if you prefer :) |
| 00:28.56 | sibiria | i think you can only access the CDR values you Set via cdr_custom |
| 00:30.15 | file | dst is a read-only variable |
| 00:30.36 | hdon | file, ahhhh thanks |
| 00:32.27 | *** part/#asterisk kharwell (kharwell@nat/digium/x-gmofohgpcnctrari) |
| 00:38.09 | hdon | what does the forward slash do in dialplan when specifying extension? e.g. exten => _NXXNXXXXXX/_12XX,1,SomeApp() |
| 00:38.54 | hdon | ohh... caller id matching! |
| 00:39.44 | TandyUK | Samot: ok, and how does that help lol |
| 00:39.54 | TandyUK | i think you totally and utterly miss the point of my 'problem' |
| 00:40.21 | TandyUK | [2017-11-07 00:39:15] == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) |
| 00:40.21 | TandyUK | [2017-11-07 00:39:15] WARNING[5729]: tcptls.c:683 handle_tcptls_connection: FILE * open failed! |
| 00:40.22 | TandyUK | happens spread amonst about 50 different sip messages being sent or recieved |
| 00:40.27 | TandyUK | in the same second |
| 00:40.49 | TandyUK | how on earth does that tell me anything about which of those 50 sip messages it actually relates to? |
| 00:41.15 | TandyUK | sure it narrows it down a little, but not enough to actually be useful |
| 00:41.54 | file | none of the SIP messages. |
| 00:42.05 | file | if the TLS connection can't be set up, then SIP messages can't flow over it |
| 00:43.07 | TandyUK | right, so in order to stop this, how o i find out more info about what asterisk was trying to do when it threw that warning? |
| 00:43.30 | file | generally that's from OpenSSL bubbled up into Asterisk |
| 00:44.16 | file | what version of Asterisk? |
| 00:44.20 | TandyUK | 11 |
| 00:44.35 | TandyUK | .25.1 |
| 00:44.48 | file | later versions query OpenSSL deeper and in a more detailed fashion to provide information. |
| 00:46.16 | file | a packet capture would also indicate failed TLS negotiation |
| 00:49.35 | TandyUK | ok give it a few mins |
| 01:09.40 | TandyUK | well i thought i was onto something, but no lol |
| 01:10.00 | TandyUK | bbiab when ive got a much larger pcap to analyse |
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| 02:10.25 | LiuYan | Does res_odbc forbid empty/blank password ? I configure a unixODBC data source (MySQL) in /etc/odbc.ini, and use isql to test it, it works. but in asterisk, it tell me "res_odbc: Error SQLConnect=-1 errno=1698 [unixODBC][MySQL][ODBC 5.1 Driver]Access denied for user 'root'@'localhost'" when `mobule reload res_odbc` |
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| 05:34.49 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 05:34.49 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.0 (2017/10/30), 11.25.3 (2017/09/19), Standard: 15.1.0 (2017/10/30); 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 |
| 05:40.46 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 05:40.46 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.0 (2017/10/30), 11.25.3 (2017/09/19), Standard: 15.1.0 (2017/10/30); 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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| 06:39.38 | Rasputin3711 | What is the best practices to install asterisk on Hyper-V ? |
| 07:10.52 | *** join/#asterisk Bhakimi (~textual@rrcs-76-81-63-218.west.biz.rr.com) |
| 07:11.25 | Bhakimi | is it possible to dial a extension at a server and also define the login and password of the server all inside the dial function ? |
| 07:15.21 | Bhakimi | with iax i can do Dial(IAX2/username:passord@peername/extension |
| 07:17.26 | LiuYan | maybe define a SIP account as a SIP trunk, then dial through it? like Dial(SIP/TheTrunkName/the-extension) |
| 07:17.52 | Bhakimi | yea, im trying to avoid doing that |
| 07:22.34 | Bhakimi | tried ur idea |
| 07:22.38 | Bhakimi | this is what i get username mismatch, have <safesoft-dialer-trunk-2087813824>, digest has <s> |
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| 08:04.40 | LiuYan | Bhakimi: There's guide to setup SIP trunk in Asterisk, it works for me. --> https://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx |
| 08:05.04 | Bhakimi | thank you, let me read it |
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| 08:06.47 | Bhakimi | let me ask you this |
| 08:07.07 | Bhakimi | lets say i want to send calls between two different asterisk boxes back and forth |
| 08:07.27 | Bhakimi | but some calls i like to send to a specific contexts between server a and b |
| 08:07.42 | Bhakimi | and others to a different context between a and b |
| 08:07.54 | Bhakimi | what would be the best way to acomplish this? |
| 08:11.54 | LiuYan | Bhakimi: As far as I know, one SIP account/trunk is bind to one context. If you need two different context, maybe, setup two trunks between a and b ? |
| 08:12.28 | Bhakimi | thats what i did but when the calls go back and forth they dont follow the proper trunk |
| 08:13.59 | voipmonk | what distinguishes them as different calls? |
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| 08:20.16 | LiuYan | Bhakimi: can you post your dialplans of both A and B servers? |
| 08:20.49 | Bhakimi | using your url i think i just got it to work ;) |
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| 08:20.57 | Bhakimi | i was using type=peer |
| 08:21.01 | Bhakimi | and it wasnt having it |
| 08:21.14 | Bhakimi | setting it to type=friend and insecure=port,invite solved the issue |
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| 08:30.57 | LiuYan | Bhakimi: usually 'type=peer' works, you also need to register to remote server using 'register => USERNAME:PASSWORD@THE_OTHER_SERVER', it's a config item in sip.conf [general] section. |
| 08:34.44 | Bhakimi | yea i did the register part |
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| 09:51.11 | Bhakimi | LiuYan: are you still around ? |
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| 10:29.52 | GregUK | Hi guys, I'm running Asterisk 13.14.1 and appear to be hitting a limit, which is causing the system to hang: WARNING[25259] taskprocessor.c: The 'pjsip/distributor-000000c5' task processor queue reached 500 scheduled tasks. Does anyone know where this is defined please so that I can increase it? |
| 10:31.08 | file | that's not an explicit limit, it's a threshold which prints that warning |
| 10:31.34 | file | it means that the thread which handles and dispatches incoming traffic is backed up for some reason |
| 10:37.47 | GregUK | file: Thanks for clarifying. How can I locate the source of the problem? |
| 10:38.36 | file | well - fixes were done in later versions for deadlocks in that area, so it might already be resolved |
| 10:38.44 | file | otherwise you'd need to get a backtrace and identify where it is hung |
| 10:39.17 | file | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock |
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| 10:59.11 | GregUK | file: Thanks, I will look into upgrading and/or getting a backtrace. |
| 11:00.23 | GregUK | file: I've just noticed that prior to that warning, the server was subjected to a brute-force attack, so it looks like that's what caused the problem. |
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| 11:36.06 | bounceman | Howdy, I need some guideance. Our asterisk has gone crazy. It does not seem to start up properly. If running asterisk -vvvvvvc it stops at == Parsing '/etc/asterisk/cli.conf': Found |
| 11:36.14 | bounceman | The next step after is "Asterisk is ready" |
| 11:36.33 | bounceman | I cannot use any CLI commands either if I start it by simply service asterisk start |
| 11:40.37 | bounceman | I'll just try to rebuild it |
| 11:50.03 | bounceman | Now it seems to start I think, but there is no output in the CLI if I type sip show peers |
| 11:50.11 | bounceman | It's completely dead |
| 11:50.22 | *** join/#asterisk Guest11377 (gruetzkopf@bnc.dont-follow.us) |
| 11:51.50 | bounceman | asterisk -rx 'sip show peers' just hang as well |
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| 11:53.51 | oroborous | Samot: is it possible to trace SIP packets? |
| 11:57.11 | bounceman | http://codepad.org/3nudhTEg |
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| 12:01.49 | DanQuinney | bounceman: malformed conf file(s)? |
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| 13:11.51 | Samot | oroborous: Please stop pinging me. |
| 13:13.06 | oroborous | Samot: like in ICMP ECHO? |
| 13:15.37 | Samot | No. Like in don't use my name in messages to get my attention. |
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| 13:51.04 | *** join/#asterisk [sID] (~sid@logv.pl) |
| 13:52.04 | [sID] | Hi, How can I set intra company route in dialplan for internal connections? |
| 13:53.10 | Samot | There's no such thing as "intra company" in Asterisk. |
| 13:53.57 | *** join/#asterisk retentiveboy (~retentive@c-73-82-30-193.hsd1.ga.comcast.net) |
| 13:55.12 | [sID] | Samot: It needs something outbound in freepbx to pass CID sip: |
| 13:56.39 | Samot | OK this is not #freepbx |
| 13:56.47 | Samot | Please stop asking #freepbx questions in here. |
| 13:56.52 | Samot | There is no GUI support here. |
| 13:57.43 | [sID] | I do not use FreePBX? |
| 13:58.09 | Samot | 8:55:13 AM <[sID]> Samot: It needs something outbound in freepbx to pass CID sip: <-- Then why did you say this? |
| 13:58.32 | Samot | "intra company" is dialplan stuff |
| 13:58.43 | Samot | Not SIP trunk settings stuff. |
| 13:59.07 | Samot | How you want to handle/setup calls between the two systems is up to you. |
| 14:01.40 | [sID] | I do not know how to change the header from: |
| 14:01.59 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 14:02.59 | Samot | What does it need to change to? |
| 14:05.04 | [sID] | From: "1000" <sip: 33 @ a need to have From: "1000" <sip: 1000 @ |
| 14:05.30 | Samot | Then set the callerid. |
| 14:06.10 | Samot | CALLERID(num) and CALLERID(nam) set the callerid at the FROM header level. |
| 14:06.26 | Samot | Then build PAI and RPID headers from that if required. |
| 14:06.41 | Samot | (num) sets the from user |
| 14:06.47 | Samot | (nam) sets the from name |
| 14:07.16 | Samot | You're just passing the original callerid of the device |
| 14:08.03 | [sID] | hmm |
| 14:09.05 | [sID] | Something like that: |
| 14:09.09 | [sID] | SipAddHeader (Remote-Party-ID: <sip: $ {CALLERID (num)} @ |
| 14:09.19 | Samot | No. |
| 14:09.21 | Samot | WTF. |
| 14:09.32 | Samot | Just SET the CALLERID |
| 14:09.47 | Samot | I didn't say add SIPHeaders. |
| 14:09.56 | [sID] | I set up callerid, but it does not work |
| 14:10.03 | voipmonk | where? |
| 14:10.06 | Samot | Well are you setting it up right? |
| 14:10.17 | Samot | Show your dialplan for this |
| 14:10.19 | Samot | ~pb |
| 14:10.20 | infobot | [pastebin] 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. |
| 14:11.13 | file | setting "fromuser" will also override the value in outgoing From... as that is what the option does |
| 14:11.17 | file | or from_user if using PJSIP |
| 14:11.25 | Samot | Right |
| 14:11.28 | Samot | But that is static |
| 14:11.34 | Samot | If you do it at the peer level |
| 14:11.41 | file | yes |
| 14:11.46 | Samot | I'm under the impression he wants to pass it per caller |
| 14:11.54 | Samot | Since it sounds like two locations for the same company |
| 14:12.00 | [sID] | https://pastebin.ca/3905909 |
| 14:12.04 | file | sure, but if he has that option set and doesn't realize that... then boom, it'll never work |
| 14:12.13 | Samot | Well that's true |
| 14:12.17 | Samot | Was going to check that |
| 14:12.25 | Samot | Once I confirmed he was sending it right to begin with |
| 14:12.54 | Samot | same => n,Set(CALLERID(name)=${CALLERID(number)}) |
| 14:12.54 | Samot | <PROTECTED> |
| 14:13.05 | Samot | Which his is completely not. :) |
| 14:13.49 | Samot | [sID]: 1) Why is device 33 needing to be sent as "1000"? |
| 14:14.03 | [sID] | I tried for num and name |
| 14:14.08 | Samot | [sID]: 1) Why is device 33 needing to be sent as "1000"? |
| 14:14.26 | [sID] | Because there is a need |
| 14:14.33 | Samot | I get that |
| 14:14.38 | Samot | I'm asking why |
| 14:14.46 | Samot | Because it's impacting what you want to do |
| 14:14.57 | [sID] | The system on the other side only in that form accepts |
| 14:15.02 | Samot | OK |
| 14:15.09 | Samot | Then you need to CHANGE the CALLERID for that call. |
| 14:15.16 | Samot | You're passing "33" as the USER |
| 14:15.37 | [sID] | 33 is trunk |
| 14:15.44 | Samot | OK |
| 14:15.51 | Samot | So now back to what file was getting to |
| 14:15.53 | [sID] | Serwer A -> Serwer B |
| 14:16.00 | Samot | Do you have the fromuser= set in the trunk? |
| 14:16.10 | [sID] | yes |
| 14:16.12 | Samot | K |
| 14:16.14 | Samot | That's wrong. |
| 14:16.18 | [sID] | hmm |
| 14:16.19 | Samot | It will ALWAYS set that. |
| 14:16.24 | Samot | That's why that setting is there. |
| 14:16.48 | [sID] | ok, it will change and I will check |
| 14:16.55 | Samot | 9:12:05 AM <@file> sure, but if he has that option set and doesn't realize that... then boom, it'll never work |
| 14:16.57 | Samot | ^^^^ |
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| 16:38.44 | *** join/#asterisk bar98 (~bar98@24.137.204.75) |
| 16:40.53 | bar98 | Hi |
| 16:42.37 | bar98 | I have an application that sends in the SDP additional Media Attributes. I need to make asterisk to pass these attributes. |
| 16:43.36 | bar98 | Can I get some directions on where to look at the code to do the changes? Can I use pjsip only for this? |
| 16:48.10 | bar98 | This is the right place to ask this question? |
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| 16:59.29 | bar98 | hello |
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| 17:09.13 | Samot | bar98: What additional attributes? |
| 17:09.30 | Samot | And Asterisk doesn't "pass" things. It's a B2BUA |
| 17:10.10 | Samot | Asterisk is going to answer the call from your application. Where it goes and if it leaves Asterisk is up to you. |
| 17:10.46 | bar98 | there are many attributes like a=rtphe:1 |
| 17:10.48 | Samot | But then any Dial() or Originate is going to create new channels with entirely new SDP |
| 17:11.50 | Samot | The two "legs" of the call have their own channels and SDP |
| 17:11.56 | Samot | Asterisk bridges them together. |
| 17:12.04 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 17:12.06 | Samot | You app is going to talk to Asterisk |
| 17:12.11 | Samot | Asterisk is going to talk to the other side. |
| 17:12.21 | Samot | Asterisk handles bridging the two sides. |
| 17:12.57 | bar98 | That is clear for me |
| 17:13.27 | Samot | file: Can you add SDP body in PJSIP for outbound calls? |
| 17:13.33 | bar98 | Asterisk could read the attributes when a call is received and use them when it creates the new SDP |
| 17:13.53 | file | you can't arbitrarily change SDP from dialplan or anything, no, it is produced outside of your control |
| 17:14.13 | Samot | Does Asterisk accept any SDP body? |
| 17:14.20 | bar98 | yes |
| 17:14.31 | Samot | But on his new channel he can't add it. |
| 17:14.38 | bar98 | correct |
| 17:14.43 | file | it doesn't accept any SDP body |
| 17:14.45 | file | it is parsed, negotiated |
| 17:14.50 | file | some parts are ignored if we don't know about them |
| 17:14.58 | Samot | Sorry, you both have the same nick color |
| 17:15.04 | Samot | My questions were for file. |
| 17:15.10 | Samot | bar98: There's your answer. |
| 17:15.20 | bar98 | well it doesnt reject the additional attributes |
| 17:15.23 | Samot | Asterisk will not accept SDP body it doesn't know about |
| 17:15.44 | Samot | Asterisk will not let you add or interject SDP body. |
| 17:16.19 | bar98 | when I use pjsip all the SDP handling is done by pjsip? |
| 17:17.22 | file | the res_pjsip_sdp_rtp.c module handles the RTP part of SDP, and res_pjsip_session.c handles some of the core SDP stuff, and the PJSIP negotiator negotiates it some |
| 17:17.49 | Samot | bar98: This is not really something Asterisk is meant to do. |
| 17:17.53 | Samot | It's a B2BUA |
| 17:17.57 | Samot | Not an SBC or a Proxy |
| 17:18.52 | Samot | The ability to read/manipulate the SIP Message itself is very limited in Asterisk. |
| 17:19.05 | bar98 | with some code changes in pjsip can I achieve what I need ? ot there are more code outside pjsip that deals with SDP ? |
| 17:19.20 | *** join/#asterisk aness (~aness@178.74.21.61) |
| 17:19.52 | file | Asterisk wasn't designed to transport arbitrary information like that across things, you have to modify more than PJSIP |
| 17:20.12 | Samot | 12:17:53 PM <Samot> It's a B2BUA |
| 17:20.12 | Samot | 12:17:58 PM <Samot> Not an SBC or a Proxy |
| 17:20.12 | Samot | 12:18:52 PM <Samot> The ability to read/manipulate the SIP Message itself is very limited in Asterisk. |
| 17:20.19 | Samot | In case you missed it. |
| 17:20.28 | Samot | You need an SBC to do this stuff. |
| 17:20.39 | Samot | Asterisk is not designed or meant for what you want to do with the message. |
| 17:20.52 | lvlinux | bar98: look into Kamailio |
| 17:20.55 | Samot | A B2BUA is not something for this. |
| 17:20.57 | *** join/#asterisk tripleslash (~triplesla@unaffiliated/imsaguy) |
| 17:20.59 | Samot | Well.. |
| 17:21.08 | Samot | It depends on if he needs the media handled. |
| 17:21.13 | Samot | Kamailio doesn't do media. |
| 17:21.41 | lvlinux | true |
| 17:22.10 | Samot | Modifying of the SDP body, sure that could be done there. |
| 17:22.20 | Samot | But then it's endpoint to endpoint for the media. |
| 17:22.24 | bar98 | Kamailio works by himself or you need also asterisk ? |
| 17:22.46 | Samot | What is this application doing? |
| 17:23.18 | Samot | It generates a call and sends it to Asterisk? |
| 17:23.21 | bar98 | The application talks to radios which supports ed-137A |
| 17:23.30 | Samot | And then Asterisk sends it where? |
| 17:23.34 | *** join/#asterisk retentiveboy (~retentive@c-73-82-30-193.hsd1.ga.comcast.net) |
| 17:24.48 | bar98 | I want asterisk between the application and the radios |
| 17:25.03 | Samot | Because? |
| 17:25.09 | Samot | What is the need for it? |
| 17:25.09 | bar98 | becuase asterisk also could access the telephone lines |
| 17:25.32 | Samot | So you need to connect to FXO/FXS? |
| 17:25.38 | bar98 | correct |
| 17:25.54 | Samot | OK so.. |
| 17:26.06 | Samot | You cannot modify the SDP in Asterisk. |
| 17:26.16 | Samot | Again, it's a B2BUA |
| 17:26.25 | Samot | The radios are talking to Asterisk. That's it. |
| 17:26.37 | Samot | They don't know about the application. |
| 17:26.43 | Samot | The same is for the application. |
| 17:26.53 | bar98 | that is clear for me. |
| 17:27.00 | Samot | Your SDP between the Application and Asterisk... |
| 17:27.11 | Samot | Has not bearing on the SDP between Asterisk and the Radios. |
| 17:27.32 | Samot | So there is no "passing" |
| 17:27.45 | Samot | Because there is no direct SDP between the application and the radios. |
| 17:28.06 | bar98 | how Kamailio can help me with this ? |
| 17:28.24 | Samot | It really can't. |
| 17:28.37 | Samot | Why does the radio need these SDP settings? |
| 17:28.37 | bar98 | Asterisk can handle the Digium E1 card that have haccess to the radios |
| 17:29.07 | bar98 | it is part of the ED-137A which is used in the aviation. |
| 17:29.21 | Samot | OK |
| 17:29.29 | Samot | This is probably not the PBX for you then. |
| 17:29.39 | Samot | You need something that can work with your needs more. |
| 17:29.53 | Samot | If this is aviation, there are systems out there I'm sure. |
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| 17:32.05 | bar98 | Kamailio is a proxy and could change the SDP but once it connects to Asterisk the SDP is not changed any more. |
| 17:32.30 | Samot | Kamailio doesn't do anything with the media. |
| 17:32.32 | bar98 | Also I need a switch that handle the Digium card |
| 17:32.45 | Samot | IT can add / remove stuff from the SIP body include SDP body content |
| 17:33.02 | bar98 | as far as I know only Asterisk works with Digium boards, correct? |
| 17:33.31 | Samot | That's an E1 |
| 17:33.35 | bar98 | correct |
| 17:33.41 | Samot | Those codecs go out the window |
| 17:33.56 | bar98 | which codecs |
| 17:34.18 | Samot | How does Asterisk talk to the radios? |
| 17:34.27 | Samot | Is it via SIP or through the E1? |
| 17:34.29 | file | if you're connecting to the PSTN using a card and chan_dahdi then what would the information from the SDP even be used for |
| 17:34.33 | bar98 | via SIP |
| 17:34.53 | Samot | Where does the E1 come in? |
| 17:35.38 | bar98 | sorry E1 is only for FXO/FXS |
| 17:35.51 | Samot | If this is App -> (SIP) -> Asterisk -> (SIP) -> Radios |
| 17:35.57 | bar98 | correct |
| 17:35.58 | Samot | What is the E1 used for? |
| 17:36.06 | Samot | Why do you need it? |
| 17:36.23 | bar98 | to access the FXO/FXS that are on a databank |
| 17:36.39 | Samot | For what reason? |
| 17:36.46 | Samot | This is changing scope now. |
| 17:36.52 | bar98 | they are telephone lines |
| 17:36.58 | Samot | Now we're just not talking to the Radios |
| 17:37.03 | Samot | We're talking to POTS lines. |
| 17:37.04 | Samot | Why? |
| 17:37.19 | Samot | What is the need for this for the App to talk to the Radios via Asterisk? |
| 17:37.22 | bar98 | the app needs to acess radios and telephone lines |
| 17:37.28 | Samot | OK. |
| 17:37.56 | Samot | Then the app may need to talk to two different systems. |
| 17:38.14 | Samot | Asterisk cannot handled the SDP needs of the Radios. |
| 17:38.23 | bar98 | ok, thanks |
| 17:39.03 | lvlinux | A few other systems can use Digium cards. |
| 17:39.16 | bar98 | can you name them ? |
| 17:39.53 | Samot | If the app is using SIP it could just send those to a FXO gateway |
| 17:40.06 | Samot | Which plugs into the phone line bank. |
| 17:40.18 | Samot | Asterisk is totally not needed for that |
| 17:40.34 | lvlinux | bar98: yate, freeswitch, probably some others. |
| 17:41.01 | bar98 | my databank handles FXO/FXS/E&M and more |
| 17:41.10 | lvlinux | You may want to look into yate, as I believe it does support some changing of SIP messages. I'm not sure to what extent, but it might meet your needs. |
| 17:41.14 | Samot | OK. |
| 17:41.21 | Samot | That doesn't mean you need a PBX |
| 17:41.28 | lvlinux | And Yate also handles media and other stuff as well. |
| 17:41.37 | Samot | There are devices out there that will convert SIP to various hand offs. |
| 17:41.43 | lvlinux | yes |
| 17:43.02 | bar98 | Probably I will leave the app to talk directly to the radios and for telephone use asterisk. |
| 17:44.27 | bar98 | thanks guys for your comments |
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| 18:04.29 | Samot | Why do people insist they can make Asterisk do something its not designed to do with some code changes? |
| 18:05.01 | [TK]D-Fender | Isn't that statement automatically true? |
| 18:05.02 | Samot | I mean like "I can just change some settings or add a line right?" |
| 18:05.36 | Samot | Not the "Yes you can if you change it from the core out" part |
| 18:05.45 | [TK]D-Fender | Actually.. you CAN just add your own code an compile and * will act differently accordingly. That's the entire point of OSS :) |
| 18:06.02 | [TK]D-Fender | Now if you want to go through whatever work that will entiail.. more power to you! |
| 18:06.07 | [TK]D-Fender | and ... |
| 18:06.10 | [TK]D-Fender | ~wglwat |
| 18:06.10 | infobot | wglwat is, like, well, good luck with all that |
| 18:06.19 | Samot | Thats what i mean.. |
| 18:06.41 | Samot | The impressions some quick code changes are going to make it magically work |
| 18:06.50 | Samot | And the we should know what to change |
| 18:07.09 | Samot | After being told its not the right answer |
| 18:07.27 | Samot | A B2BUA isnt the answer to talk to the radios |
| 18:08.47 | [TK]D-Fender | probably not a solution that will feel smart or efficient in terms of what it'd take.... |
| 18:08.53 | Samot | I know it is crazy but sometimes Asterisk and/or FOSS isnt the answer |
| 18:09.12 | [TK]D-Fender | or not worth the effort for the goal at hand... |
| 18:09.13 | file | I'd modify that |
| 18:09.29 | file | it isn't the answer if you don't have the knowledge, time, and patience to make it fit |
| 18:09.47 | Samot | Or |
| 18:09.58 | Samot | Its not fiscal |
| 18:09.59 | [TK]D-Fender | Shitty solutions .. are still solutions! |
| 18:10.15 | [TK]D-Fender | (Bbrought to you by our sponsor: Masochism!) |
| 18:10.24 | Samot | Technical doesnt have to be why |
| 18:11.12 | *** join/#asterisk dar123 (~dar@2602:306:bcbf:e750:5000:bee5:542e:5086) |
| 18:11.40 | Samot | FOSS comes with costs |
| 18:12.23 | Samot | How those costs weigh against a non-FOSS cost is subjective |
| 18:12.45 | Samot | Cost not just meaning $$ |
| 18:15.20 | *** join/#asterisk tzafrir (~tzafrir@62-90-199-170.barak.net.il) |
| 18:15.42 | dar123 | hey guyz, i am new to asterisk. I am trying to enable a call_fwding asterisk. Cellphone -> Provider1 -> Asterisk -> Provider2 |
| 18:17.00 | [TK]D-Fender | how is that "call forwarding"? |
| 18:17.30 | [TK]D-Fender | So far you describe a call coming to your server and then at soem point a call out. |
| 18:17.38 | [TK]D-Fender | what makes that "forwarding"? |
| 18:17.46 | voipmonk | burns all the Halloween candy |
| 18:17.57 | Samot | Just change the code. |
| 18:18.04 | Samot | call_fwding = true; |
| 18:18.06 | Samot | Done. |
| 18:18.22 | [TK]D-Fender | match your inbound call. Dial() out. The end. |
| 18:18.25 | Samot | It's right next to asterisk_cluster = enabled |
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| 19:44.14 | jamesaxl | hi |
| 19:45.05 | jamesaxl | could you help me about Monitor function, I wonder why can not mv the recording file from monitoring to another folder https://pastebin.com/nYMFqJ6b |
| 19:46.05 | [TK]D-Fender | Why should you be able to? |
| 19:46.38 | [TK]D-Fender | You told it to START recording .. wheat makes you think you can rename a file while it's being WRITTEN TO? |
| 19:48.46 | MrTAP | [TK]D-Fender: actually, on linux you can rename files while they're being written to. the file handle remains open and doesn't care whatsoever. but jamesaxl that seems like a great way to have yourself a race condition |
| 19:49.20 | MrTAP | why not just name the file correctly to begin with? |
| 19:49.28 | [TK]D-Fender | News to me... |
| 19:50.44 | jamesaxl | MrTAP: I see that Monitor Add suffix -in or -out |
| 19:51.18 | jamesaxl | MrTAP: if I rename the file jamesaxl it will be automaticaly jamesaxl-in.wav or jamesaxl-out |
| 19:52.10 | MrTAP | set MONITOR_EXEC and rename them after the recording is done |
| 19:52.24 | MrTAP | or use MixMonitor instead |
| 19:52.50 | jamesaxl | MrTAP: I am going to try MixMonitor then, thank you |
| 19:53.23 | jamesaxl | [TK]D-Fender: do you have a Fender? |
| 19:54.13 | [TK]D-Fender | no, nick was not music related |
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| 20:15.45 | jamesaxl | is it possible to get the id of operator when (s)he has a call in queue? |
| 20:17.37 | [TK]D-Fender | who is this "operator"? |
| 20:17.49 | [TK]D-Fender | And how does an individual have a call in a queue? |
| 20:19.08 | Samot | Words. Who need |
| 20:20.08 | jamesaxl | [TK]D-Fender: It is an agent that belong queue support, and queue support has many agents, I would like to get the sip id of every egent that took the call in support queue |
| 20:20.23 | [TK]D-Fender | too THE call? |
| 20:20.31 | [TK]D-Fender | you don't get multiple people talking ONE call |
| 20:20.43 | jamesaxl | [TK]D-Fender: s/too/to |
| 20:20.54 | jamesaxl | [TK]D-Fender: of course nop :) |
| 20:21.11 | [TK]D-Fender | YThe more you explain the more you break the concept of "explaining" |
| 20:22.18 | jamesaxl | [TK]D-Fender: e.g: you belong to queue support and I want to get your sip id when you have a incoming call, that is all |
| 20:22.33 | [TK]D-Fender | huh? |
| 20:22.36 | jamesaxl | [TK]D-Fender: what is your native language please ? |
| 20:22.43 | [TK]D-Fender | ENGLISH |
| 20:23.07 | jamesaxl | does believe |
| 20:23.14 | [TK]D-Fender | What "sip ID" are you even talking about? |
| 20:23.14 | jamesaxl | doesn't believe |
| 20:23.26 | [TK]D-Fender | You are using VAGUE SHIT TERM across the board. |
| 20:23.40 | jamesaxl | [TK]D-Fender: this for example [jamesaxl] |
| 20:23.43 | Samot | OK |
| 20:23.44 | [TK]D-Fender | "I would like to get the sip id of every egent that took the call in support queue" <- multiple agents. ONE call. |
| 20:23.45 | Samot | First.. |
| 20:23.50 | [TK]D-Fender | what. the. FUCK? |
| 20:23.55 | Samot | Queues show what agents are on calls.. |
| 20:24.00 | Samot | Is that what you want to know? |
| 20:24.07 | Samot | And what channels those calls are on.. |
| 20:24.10 | [TK]D-Fender | <jamesaxl> [TK]D-Fender: this for example [jamesaxl] <- that's a PEER |
| 20:24.25 | [TK]D-Fender | and WHERE do you expect to get your hands on this? |
| 20:24.32 | Samot | jamesaxl: There is already stats for this. |
| 20:25.14 | Samot | The queues will show agents and their status |
| 20:25.16 | jamesaxl | Samot: I knew that, I want to know how to get it, with using exten |
| 20:25.24 | Samot | No. |
| 20:25.30 | [TK]D-Fender | get it WHEN? |
| 20:25.32 | Samot | You look at the queue or its log. |
| 20:25.52 | [TK]D-Fender | get it WHERE? |
| 20:26.49 | jamesaxl | Samot: i just want to put the agent (peer or extension) when recording file that is all |
| 20:27.37 | Samot | So you should be getting that information when the agent answers or somethhing |
| 20:27.42 | [TK]D-Fender | ^ |
| 20:27.44 | Samot | Before you name the file. |
| 20:27.55 | [TK]D-Fender | "core show application queue" <- |
| 20:27.57 | Samot | Name the file properly at the start. |
| 20:28.03 | [TK]D-Fender | And look at how you call your members |
| 20:28.13 | Samot | And how to handle a call that is answered |
| 20:28.17 | Samot | Before it bridges. |
| 20:28.21 | [TK]D-Fender | yup |
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| 20:28.52 | jamesaxl | understood, thank you both :) |
| 20:31.17 | jamesaxl | I see some people using Freepbx ? do you suggest it ? |
| 20:32.35 | [TK]D-Fender | Does it fit your needs? |
| 20:32.58 | [TK]D-Fender | Any suggestion without a proper description of needs is pointless |
| 20:33.03 | jamesaxl | [TK]D-Fender: nop, because it is close |
| 20:33.09 | [TK]D-Fender | Do I recommend a For F350? |
| 20:33.18 | [TK]D-Fender | Ford* |
| 20:33.24 | [TK]D-Fender | maybe ... FOR WHOM? |
| 20:34.21 | jamesaxl | [TK]D-Fender: for a huge call center with more then 20 projects |
| 20:34.32 | [TK]D-Fender | that doesn't mean anything |
| 20:34.42 | Samot | FreePBX is just a GUI |
| 20:34.45 | Samot | Over Asterisk. |
| 20:34.57 | Samot | It has call center add-ons |
| 20:35.18 | Samot | But outside of that if you need reporting look at AsterNIC |
| 20:35.22 | jamesaxl | [TK]D-Fender: I do not want to speak more about how to manage queues etc, you can continue with your mind if I say a huge call center |
| 20:35.29 | [TK]D-Fender | It has canned-call-flows. Things you may agree with. Things you may not. |
| 20:35.46 | Samot | We don't know if it will fit your needs. |
| 20:35.49 | [TK]D-Fender | ^ |
| 20:36.06 | Samot | It's pretty customizable over all |
| 20:36.26 | Samot | But only you can make the decision if it is something that would work for you. |
| 20:36.31 | [TK]D-Fender | Huge call center can have BORING fucking needs with NOTHiNG special where you want some dumb chimp employee to add "users" with a point& click GUI because they have no technical knowledge and maybe that's good for YOU. |
| 20:36.35 | [TK]D-Fender | WE DON'T KNOW WHAT |
| 20:36.37 | jamesaxl | I create my own realtime monitoring :) with QT5/c++ and Perl/mojolicious as an API |
| 20:36.49 | [TK]D-Fender | So just saying "huge call center" mean NOTHING |
| 20:37.00 | jamesaxl | I just need to test it more and more and I can share it under BSD licence |
| 20:37.04 | [TK]D-Fender | What was I saying about "vague shit terms"? |
| 20:37.17 | Samot | Test it more? |
| 20:37.23 | Samot | You need to understand how Asterisk handles queues |
| 20:37.27 | [TK]D-Fender | If it good for 50 people? Not if the processing it generates doesn't do what you WANT. |
| 20:37.28 | Samot | And how to use those queues |
| 20:37.31 | Samot | Before you can "test it more" |
| 20:37.46 | [TK]D-Fender | Quantity has NOTHING to do with it |
| 20:37.51 | Samot | ^^^^ |
| 20:38.09 | [TK]D-Fender | HOW does it work. WHOSE life does it make better? Who has to MANAGE it? |
| 20:38.36 | Samot | Does it compare to the other, viable and known options? |
| 20:39.02 | jamesaxl | Do you know Mathias? |
| 20:39.22 | [TK]D-Fender | Which one? |
| 20:39.31 | Samot | A person? |
| 20:39.31 | [TK]D-Fender | It's a pretty common first name |
| 20:39.33 | Samot | A company? |
| 20:39.39 | Samot | A program? |
| 20:40.00 | Samot | Do you know Gary? |
| 20:40.16 | [TK]D-Fender | I know Peter, Paul, and Mary. Is that close enough? |
| 20:41.07 | jamesaxl | [TK]D-Fender: pascom guy :) |
| 20:41.35 | Samot | OK.. |
| 20:41.42 | [TK]D-Fender | And am I somehow supposed to know what this "pascom" you're referring to is exactly too? |
| 20:41.55 | voipmonk | secret passwords |
| 20:42.13 | Samot | Why is it that people say they are working on complicated custom stuff for Asterisk but then can't relay the basic details? |
| 20:42.27 | *** join/#asterisk rangotail (~rangotail@unaffiliated/rangotail) |
| 20:42.39 | Samot | Or even more to the core of it.. |
| 20:42.43 | [TK]D-Fender | wonders why so many people coming to a telecommunications channel CAN'T FUCKING COMMUNICATE. |
| 20:42.50 | Samot | Understand the basics of what Asterisk does. |
| 20:42.55 | Samot | Right. |
| 20:44.21 | jamesaxl | [TK]D-Fender: because of you |
| 20:44.33 | Samot | Little kids cry |
| 20:44.48 | Samot | Sorry, was I not supposed to finish that sentence? |
| 20:46.47 | Samot | Because it seems you're not... |
| 20:47.13 | jamesaxl | I want also to check outgoing calls but from queue, is it possible to get the queue member that doing an outgoing call? |
| 20:47.37 | Samot | Queues doesn't make outbound calls.. |
| 20:47.41 | jamesaxl | I know that I have the possibility to check CDR |
| 20:47.50 | Samot | Outside of calling from Asterisk -> Device |
| 20:48.10 | Samot | No one makes "outbound" calls to the PSTN through queues. |
| 20:48.18 | Samot | They make them from their device. |
| 20:48.19 | *** join/#asterisk jkroon (~jkroon@165.16.204.167) |
| 20:48.22 | [TK]D-Fender | #wordsoup |
| 20:48.39 | [TK]D-Fender | adds some paprika and stirs a bit more |
| 20:48.41 | Samot | Device sends call to Asterisk.. |
| 20:48.49 | Samot | Asterisk puts call in device's context. |
| 20:48.59 | Samot | Digits are matched |
| 20:49.03 | Samot | Something happnes. |
| 20:49.21 | Samot | How a queue plays into that, I'm not sure. |
| 20:49.55 | jamesaxl | Samot: I know that :( , I just do not want to create a VIEW in data base and filter agents that doing outbound and belong to a specific queue |
| 20:50.22 | Samot | Outbound calls aren't part of a queue |
| 20:50.33 | Samot | That's agent monitoring |
| 20:50.39 | Samot | Not queue monitoring |
| 20:50.46 | [TK]D-Fender | not even "agent". |
| 20:50.49 | Samot | Two different sub-sets |
| 20:50.50 | Samot | "device" |
| 20:50.53 | [TK]D-Fender | ^ |
| 20:50.57 | Samot | That is linked to agent |
| 20:51.04 | [TK]D-Fender | depending... |
| 20:51.12 | Samot | Logged in or static |
| 20:51.14 | [TK]D-Fender | considering chan_agent is pretty much dead.... |
| 20:51.16 | Samot | It's "linked" |
| 20:51.20 | Samot | User/Device |
| 20:51.21 | jamesaxl | I understand and It is better to get info directly from CDR |
| 20:51.22 | [TK]D-Fender | and most of the time not-applicable |
| 20:51.29 | [TK]D-Fender | queues typically just have MEMBERS. |
| 20:51.33 | Samot | Yes. |
| 20:51.33 | [TK]D-Fender | that doesn't make them magical in any way |
| 20:51.50 | Samot | I'm referring to the term as in what a "reporting" software would. |
| 20:51.54 | Samot | Queues/Agents |
| 20:52.02 | [TK]D-Fender | so even most use of the term "agent" is just more vagueness added in |
| 20:52.10 | Samot | OK. |
| 20:52.20 | Samot | We all know what it means in this context. |
| 20:52.26 | Samot | It's implied. |
| 20:52.36 | jamesaxl | how about Call tracking with asterisk? any suggest please |
| 20:52.42 | Samot | I just said.. |
| 20:52.50 | [TK]D-Fender | the fact of some device being potentially called by a queue (possibly through like 4 intermediary steps) has NO relation to anything else the device may find itself involved in |
| 20:52.53 | Samot | Look at AsterNIC |
| 20:53.07 | [TK]D-Fender | define "tracking". |
| 20:53.15 | [TK]D-Fender | is CDR it? |
| 20:54.00 | [TK]D-Fender | <Samot> It's implied. <- nothing is implied when it comes from a guy who can't describe shit. Everything starts deserving scrutiny. |
| 20:54.08 | [TK]D-Fender | NOW I sound jaded.... |
| 20:54.13 | Samot | Dude. |
| 20:54.16 | Samot | I'm talking about me. |
| 20:54.21 | [TK]D-Fender | oh... |
| 20:54.23 | jamesaxl | [TK]D-Fender: get a phone leads statistic, analyze and optimize your marketing campaigns for improving efficiency. |
| 20:54.26 | Samot | When you and I are in a conversation about shit... |
| 20:54.27 | [TK]D-Fender | well... just a LITTLE for you then :p |
| 20:54.38 | Samot | It's implied that I know WTF the actual term is. |
| 20:54.46 | Samot | But after 22 years of customer support.. |
| 20:55.00 | Samot | I tend to shape the conversation in to terms they can f'ing understand |
| 20:55.07 | [TK]D-Fender | Samot, yeah I can afford you a fair amount of credit :) |
| 20:55.13 | Samot | And not focus on the minutia of the tech terms |
| 20:55.40 | [TK]D-Fender | well here NONE of the terms match. Can't communicate with something who has no common dictionary. |
| 20:55.56 | [TK]D-Fender | it becomes an uphill battle liable to end in justifiable homicde |
| 20:56.05 | [TK]D-Fender | someone* |
| 20:56.06 | [TK]D-Fender | gah... |
| 20:56.13 | Samot | When people in companies talk about "queues" and the users.. |
| 20:56.21 | Samot | The users are the "Agents" |
| 20:56.26 | Samot | "Which AGent is on call?" |
| 20:56.36 | Samot | "Which Agent is in this queue" |
| 20:56.51 | Samot | They don't sit around and go "Oh well it's a device and a user..so let's all use that properly now people" |
| 20:57.11 | Samot | Sometimes the topic of the conversation implies how the words in the conversation can be used. |
| 20:57.31 | Samot | And using common words that people can relate to, make it so much more easier. |
| 20:58.16 | [TK]D-Fender | You want to know who's answering queue call? queue.log. Want to know who's placing outbound calls not related to queue? CDR. Want something more?> go find a place to create manual logging yourself in the steps your call follows along those paths or based on external notifications like AMI <_ |
| 20:58.17 | [TK]D-Fender | There |
| 20:58.38 | Samot | That's pretty much the base of it |
| 20:58.51 | *** join/#asterisk RovingWriter (~RovingWri@unaffiliated/rovingwriter) |
| 20:59.00 | Samot | If you're writing a real-time monitoring program.. |
| 20:59.10 | Samot | You should really understand how the base works |
| 20:59.43 | [TK]D-Fender | It would be nice.... |
| 20:59.47 | Samot | This isn't something where you learn how to do X today so tomorrow you're writing a program. |
| 21:00.18 | jamesaxl | [TK]D-Fender: but with AMI how can I get abandon, unanswered calls for outgoing calls? |
| 21:00.33 | [TK]D-Fender | did yo LOOK at the even list? |
| 21:00.42 | [TK]D-Fender | enevt* |
| 21:00.48 | Samot | Event |
| 21:00.53 | [TK]D-Fender | yeah, that |
| 21:00.57 | [TK]D-Fender | asklj;dhuskdjfhlaksdjfhoasdf |
| 21:01.11 | Samot | jamesaxl: You really need to learn Asterisk first. |
| 21:01.16 | jamesaxl | [TK]D-Fender: of course, I did |
| 21:01.26 | [TK]D-Fender | Well then you'd know |
| 21:01.48 | [TK]D-Fender | AMI throws out events say a channel stats, a call is placed, a call is bridged,etc |
| 21:01.51 | [TK]D-Fender | it's ALL RIGHT THERE |
| 21:01.58 | jamesaxl | Samot: I learn it, I have this basic questions to gain more information |
| 21:01.58 | [TK]D-Fender | How is this question if you actually looked? |
| 21:02.05 | Samot | No... |
| 21:02.12 | [TK]D-Fender | Yous hould have SEEN this stuff thrown out by calls if you actually looked & tested |
| 21:02.17 | Samot | These are basic "I haven't tried" |
| 21:02.21 | [TK]D-Fender | Without even reading a command & event list |
| 21:02.23 | Samot | OR looked at the wiki |
| 21:02.25 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 21:02.38 | Samot | First off |
| 21:03.03 | Samot | Outgoing calls are not abandoned really |
| 21:03.15 | Samot | They are either answered or not. |
| 21:03.27 | Samot | They get a REPLY |
| 21:03.42 | [TK]D-Fender | well.. the caller COULD just hang up before answer |
| 21:03.48 | [TK]D-Fender | that could be an "abandon" |
| 21:03.49 | Samot | You see now.. |
| 21:03.52 | jamesaxl | Samot: I am using this https://metacpan.org/pod/Asterisk::AMI for my new API |
| 21:03.57 | Samot | Now this is where terms are important. |
| 21:04.04 | [TK]D-Fender | noboday cares about that API |
| 21:04.12 | [TK]D-Fender | we aren't limited by whatever module that is |
| 21:04.19 | Samot | An "Abandoned" call in Queues....is when the CALLER into the QUEUE leaves the queue. |
| 21:04.20 | Samot | Hangs up |
| 21:04.22 | Samot | Gets out |
| 21:04.26 | Samot | Never answered. |
| 21:04.33 | Samot | Totally different |
| 21:04.42 | Samot | Stop confusing INBOUND with OUTBOUND |
| 21:04.49 | [TK]D-Fender | Yes, and for outbound calls the caller can just hang up. Nominally different, but similar |
| 21:04.58 | [TK]D-Fender | they guy trying to get somewhere decided to stop |
| 21:05.22 | Samot | Yes but an Abandoned call in a queue is an ANSWERED called |
| 21:05.29 | Samot | Or a HELD call in the queue |
| 21:05.30 | [TK]D-Fender | on outbound the callee never asked for it, so the caller is the one 'abandoning the attempt before hitting some other failure |
| 21:05.37 | Samot | Waiting for an open member.. |
| 21:05.41 | [TK]D-Fender | it is a little greyer, but mostly valid |
| 21:05.48 | Samot | If it hangs up before getting to a member..that's abandoned.. |
| 21:05.58 | Samot | An outgoing call that is answered and then hung up |
| 21:06.02 | Samot | Isn't abandoned |
| 21:06.08 | Samot | It was just an f'ing call |
| 21:06.15 | [TK]D-Fender | more or less. |
| 21:06.32 | Samot | So in the terms of writing a queue monitor/reporting program.. |
| 21:06.34 | [TK]D-Fender | like I said it's a close parallel on the idea of hwo can "give up" |
| 21:06.41 | [TK]D-Fender | who* |
| 21:06.45 | Samot | "Abandoned Outbound" sounds lames. |
| 21:07.04 | [TK]D-Fender | lame is quickly becoming a efature of the system.... |
| 21:07.14 | [TK]D-Fender | just like my typos today |
| 21:07.22 | [TK]D-Fender | misses his mechanical keyboard.... |
| 21:07.39 | [TK]D-Fender | plays the blue-key blues.... |
| 21:07.54 | [TK]D-Fender | clickity-clickity-clickity-clickity-clickity-clickity-clickity-clickity-clickity |
| 21:19.21 | *** join/#asterisk tripleslash (~triplesla@unaffiliated/imsaguy) |
| 21:25.08 | jamesaxl | [TK]D-Fender Samot https://www.youtube.com/watch?v=C98sOdWjtsw I have only this video to thank you |
| 21:26.19 | *** join/#asterisk aness (~aness@cm-84.209.56.255.getinternet.no) |
| 21:33.12 | [TK]D-Fender | packs up to head home... |
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| 22:12.58 | *** join/#asterisk snake38 (~root@181.120.191.236) |
| 22:13.03 | snake38 | hi |
| 22:13.40 | snake38 | I'm curious if asterisk fixed the issues that lead to the creation of the freeswitch project? |
| 22:14.07 | snake38 | I'm curious how asterisk has evolved since then |
| 22:17.58 | snake38 | are there still issues with the channel creation and cloning the channel, etc? please don't take this the wrong way, I'm curious |
| 22:22.28 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 22:22.30 | [TK]D-Fender | that was more than half a decade ago |
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| 22:24.22 | snake38 | I see, so I take it's no longer an issue? |
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| 22:27.42 | file | you'd have to be specific about "issues", but in a general sense Asterisk has changed quite a lot |
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| 22:30.42 | snake38 | file: I'm talking about the criticism pointed out by FS devs here: https://freeswitch.org/how-does-freeswitch-compare-to-asterisk/ /* XXX This is a seriously wacked out operation. We're essentially putting the guts of the clone channel into the original channel. Start by killing off the original channel's backend. I'm not sure we're going to keep this function, because while the features are nice, the cost is very high in terms of pure nastiness. XXX */ |
| 22:31.12 | file | the operation still exists, it's use has been minimized, and it's never seen outside of the core any longer |
| 22:31.32 | file | it's become an internal implementation detail that doesn't matter to users |
| 22:31.44 | snake38 | I see |
| 22:32.00 | snake38 | ok, I'll give the latest version a try, thanks |
| 22:32.54 | Samot | hehe. |
| 22:33.38 | Samot | 5:22:31 PM T<[TK]D-Fender> that was more than half a decade ago <-- More like almost a decade. |
| 22:33.43 | Samot | April 2008 |
| 22:33.50 | [TK]D-Fender | that far? |
| 22:34.02 | Samot | Yes, that article is from April 2008 |
| 22:34.04 | [TK]D-Fender | also more than half a decade... |
| 22:34.17 | Samot | More precisely almost a decade |
| 22:36.00 | Samot | The article is almost irrelevant now. |
| 22:38.16 | Samot | Like the fact that was close to or around the time the kernel update in Linux allowed for conferences not to require DAHDI Time. |
| 22:44.21 | snake38 | yes, that was one of their selling points, that their conference module didn't require compiling a kernel module or something like that |
| 22:44.49 | Samot | Well you're referencing a 10 year old article basically |
| 22:44.57 | Samot | It has not application to today |
| 22:45.06 | Samot | As file said things have changed a lot. |
| 22:45.14 | snake38 | I see, nice |
| 22:45.26 | Samot | That's like 1.6 era |
| 22:45.31 | Samot | aka v6 |
| 22:45.33 | snake38 | excited to try the latest version and see for myself |
| 22:45.36 | Samot | We're on 13 and 15 |
| 22:45.42 | snake38 | nice |
| 22:45.59 | Samot | Major things have happened since then. |
| 22:46.45 | snake38 | where can I see a list of things that changed? |
| 22:47.39 | Samot | The change logs. |
| 22:47.45 | snake38 | thanks |
| 22:47.46 | file | every version has a Changelog, every major release has a general list on the wiki |
| 22:47.49 | Samot | But that is quite a long list. |
| 22:48.16 | Samot | Because lets be honest.. |
| 22:48.27 | Samot | FreeSWITCHes first couple releases weren't all polished. |
| 22:48.40 | Samot | So if you want to compare then to now.. |
| 22:51.03 | [TK]D-Fender | It also isn't pure switch or pure "VoIP" either as I recall. Just another B2BUA |
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| 22:52.55 | Samot | It's not a softswitch. |
| 22:53.10 | Samot | Softswitches do not have subscribers. |
| 22:53.20 | Samot | Well proper class 4 ones |
| 22:53.37 | Samot | It's a softswitch in the sense of a class 5 switch |
| 22:53.54 | snake38 | I remember drinking their kool-aid back then, but I don't like how their platform and website has become a way to place ads for their cluecon conference, no offense |
| 22:53.57 | Samot | As in its meant to route calls between users and between a class 4 softswitch. |
| 22:54.04 | snake38 | I respect their work |
| 22:58.32 | snake38 | I also prefer asterisk being GPL licensed |
| 23:00.35 | snake38 | but yeah, I don't care about those flamewars anymore |
| 23:01.25 | snake38 | thanks y'all |
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