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| 02:42.54 | znf | raspberrypifan, key auth |
| 02:43.04 | raspberrypifan | hi |
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| 05:35.34 | raspberrypifan | is the digium certification worth it |
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| 06:08.33 | wyoung | raspberrypifan: never heard of it |
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| 08:10.59 | pawiecki | Hello! Is it possible, that dialing multiple sip peers at once (SIP/100&SIP/101&SIP102...), can cause asterisk to malfunction? |
| 08:12.57 | pawiecki | I did that for 5 peers, and phones went laggy and unreachable. When I changed it, it looks like the problem went away. Why would it make problems? |
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| 08:13.55 | pawiecki | All the phones started to dropping calls and lag, not only the 5 I dialed. |
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| 09:36.16 | Ast001 | Hello do you know some good softphone on iax2 protocol ? |
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| 11:25.42 | file | Thursday is. |
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| 12:10.29 | nixnothing | greetings |
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| 12:15.38 | nixnothing | how does asterisk compare to freeswith? |
| 12:19.26 | [TK]D-Fender | similar but different approaches. |
| 12:19.31 | [TK]D-Fender | What do you actually need? |
| 12:21.06 | nixnothing | just trying to get a feel for the differences. I'm relatively new to voip/sip projects but but just started a job that requires me to use them |
| 12:21.29 | nixnothing | I guess what I was wondering was. |
| 12:22.20 | nixnothing | we are using astersk, but I hear freeswitch might be better for what we are doing longterm b/c of database stuff |
| 12:23.05 | nixnothing | was wonding if astersk had any distint advantages that would make a difference |
| 12:23.27 | Rasputin3711 | Install freeswithc in production and compare, what is the problem? )) |
| 12:24.56 | nixnothing | no problem yet, I just dont have a strong feel for the advantages of each and havent had time to read through the differences yet in documentation |
| 12:27.03 | nixnothing | I'm told we will prob switch to freeswitch in the future from astersk b/c the databases are restricting us and we are hiting software limitations |
| 12:27.50 | nixnothing | we do cloud hosting w/ astersk |
| 12:30.01 | nixnothing | we plan to impliment some other changes as well in the future so im am just trying to get an idea of scope |
| 12:39.35 | acidfu_ | nixnothing what I like about asterisk is also the community, the number of active developers, the quantity of documentation, books, blogs, example etc |
| 12:41.22 | nixnothing | yeah, I have to agree I was impressed with the good documentation so far |
| 12:42.45 | nixnothing | I assumed the community was larger/more involved too if any just because astersk seems to be more popular |
| 12:43.02 | nixnothing | *if just |
| 12:43.09 | file | document all the things! |
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| 13:38.16 | TandyUK | gradwell having major issues atm fyi |
| 13:38.25 | TandyUK | as if anyone affected doesnt already know :P |
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| 14:16.48 | Samot | What issue is Gradwell having? |
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| 14:24.11 | TandyUK | wel ltheir IAX seems to have broken big time |
| 14:24.30 | TandyUK | theyre ackowledging outbound iax issues, btu none of our incoming iax trunks from them are recieving calls either |
| 14:24.46 | TandyUK | we just pointed the clients 0x45 numbers at a provider other than gradwell |
| 14:24.57 | TandyUK | and probably just saved our client £30/mo in the process |
| 14:25.33 | TandyUK | their tech support sent me a trace saying 'everything looks fine our end' |
| 14:25.38 | TandyUK | but the log itself shows errors lol |
| 14:25.49 | TandyUK | so they clearly havent read/understood what the trace is telling them |
| 14:26.19 | TandyUK | not to mention its a trace of _everything_ that asterisk node was doign at the time |
| 14:26.27 | TandyUK | so i now have a lot of info on soem other gradwell clients lol |
| 14:27.03 | TandyUK | im not even their client, so theyve just sent a 2mb sip debug log to some random third party (ie me) |
| 14:30.51 | file | eep |
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| 15:30.51 | jermy | Quick query - when I run 'sip show channelstats', the Recv Pack/Lost counters appear to reset to zero periodically (about 30s). Is this normal? |
| 15:34.40 | raspberrypifan | are there any new asterisk books being planned the latest one is several years old now |
| 15:38.52 | file | I do not believe so, writing such a thing takes a lot of time |
| 15:40.02 | nixnothing | also if not alot fundamentally changes, then its just waste of paper with like 5 pages off new content out of a 1000 page book |
| 15:40.56 | nixnothing | like old programming books are even more true in this way. |
| 15:41.16 | nixnothing | be easier to just look as recent changelogs with updates or version differences |
| 15:42.11 | raspberrypifan | but hasnt a lot of big stuff changed since 2013 |
| 15:42.37 | [TK]D-Fender | A few things new like PJSIP, but they're documented on the WIKI |
| 15:42.59 | [TK]D-Fender | If you feel the need to kill a tree then go and print them out |
| 15:43.20 | nixnothing | yeah asterisk seems old enough where big changes dont happen on a global level anymore |
| 15:44.04 | nixnothing | thats just new vs old software in general |
| 15:45.03 | nixnothing | its new young software with systemwide changes, updates, or redesigns that need new accross the board documentation on a regular basis (bejcause they are consolidated into 1 place) |
| 15:45.27 | nixnothing | like major version changes or revisions |
| 15:46.37 | file | user facing stuff has really only been chan_pjsip, unless you count CDRs and AMI under that umbrella |
| 15:47.12 | [TK]D-Fender | and ARI |
| 15:47.19 | file | ARI is for developers really |
| 15:47.42 | nixnothing | yeah |
| 15:49.01 | nixnothing | ARI is an asynchronous API that allows developers to build communications applications by exposing the raw primitive objects in Asterisk - channels, bridges, endpoints, media, etc. - through an intuitive REST interface. The state of the objects being controlled by the user are conveyed via JSON events over a WebSocket. |
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| 15:53.05 | *** join/#asterisk heradon (~heradon@unaffiliated/heradon) |
| 15:53.10 | heradon | hello guys |
| 15:53.15 | *** join/#asterisk Vamp898 (511b7bb0@gateway/web/cgi-irc/kiwiirc.com/ip.81.27.123.176) |
| 15:53.30 | Monpoke | Hello, I'm trying to implement a queue system and I would like to reload these queues into cli. But it seems "queue..." commands are not found... However the module "app_queue.so" is correctly loaded... Any idea? |
| 15:53.47 | Vamp898 | Hi there, small question. We use an features map for transfering calls. This works for inbound calls, but not for outbounds... we're stuck in where to look for this issue |
| 15:56.38 | Vamp898 | I choosed ## for blindxfer, if i press x |
| 15:56.58 | Vamp898 | ## in an inbound call, i can reinvite. If i press ## in an outbound call, just nothing happens (the other person hears that i press the buttons) |
| 15:57.15 | Vamp898 | not reinvite, sorry, transfer :D |
| 15:57.49 | [TK]D-Fender | Your call has to actually supportt those dial opttions |
| 15:57.59 | [TK]D-Fender | If you don't havcev them then you can't |
| 15:58.07 | [TK]D-Fender | What kind of phones are you using for those calls? |
| 15:58.31 | Vamp898 | We use the linphone app and an SIP provider for Trunking. The same provider is used for outbound and inbound calls |
| 15:59.47 | Vamp898 | If i call someone intern, the ## works, but if i call someone external (which uses the SIP provider), the ## doesnt work anymore. But with the same provider and with inbound calls, the ## works... |
| 16:01.59 | Monpoke | Problem solved |
| 16:02.00 | [TK]D-Fender | Linphone should support real transfers and not depend on DTMF through *.... |
| 16:02.23 | [TK]D-Fender | When calling out you need to set your TRUNK DIAL options to allow transfers |
| 16:02.27 | [TK]D-Fender | otherwise you won't be able to |
| 16:03.49 | Vamp898 | ah goddamit im so stupid, i set the Trunk Dial options everywhere but due to using millions of macros i extensions i forgot to set it in the very last Dial() Call which calls the provider... |
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| 16:04.32 | Vamp898 | oh wait no, actually they are set :D, sorry |
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| 16:05.48 | Vamp898 | [TK]D-Fender: dammit, now. I used trg,trgT at one dial command.... days gotta too long. Thank you for your hint, cheers! |
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| 16:06.03 | [TK]D-Fender | You're welcome |
| 16:06.14 | [TK]D-Fender | but you should still get a softphone that supports real transfers |
| 16:06.43 | [TK]D-Fender | eg: bria, Jitsi, etc |
| 16:06.51 | [TK]D-Fender | DTMF transfers = low-end crap |
| 16:07.25 | Vamp898 | [TK]D-Fender: we have that, we want to provide an backup solution for some weird phones |
| 16:07.43 | Vamp898 | [TK]D-Fender: there are other ways for linphone to transfer (nobody here uses ##), its just for other phones |
| 16:08.09 | [TK]D-Fender | Should be a few you could install that have real functionality... |
| 16:08.15 | [TK]D-Fender | But anyway just a real suggestion |
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| 16:33.06 | Synthase_ | Performed YUM update on an Asterisk 13 system using chan_sip yesterday. res_srtp had been compiled, but was unused, just for possible later use. This said, libsrtp updated yesterday and now keeps res_rtp_asterisk.so from loading, complaining of a missing shared object. I've recompiled Asterisk with the ./configure --without-srtp flag, removed libsrtp and the |
| 16:33.06 | Synthase_ | devel library, etc, no dice. Any ideas? |
| 16:35.06 | WIMPy | Did you ignore some warning about that on 'make install'? |
| 16:37.19 | Synthase_ | I stopped getting errors about during compiling after removing the packages, yet the newly created RTP module still expects it. |
| 16:37.33 | Synthase_ | Needless to say, without an RTP engine, calls are not happening. |
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| 16:39.59 | Synthase_ | While I know relinking with ld is an option with missing objects, that shouldn't be relevant when package is removed, SRTP not selected in menuselect, and using the without string during configure to force it. |
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| 16:45.37 | jrun | where did AST_CDR_FLAG_LOCKED go? |
| 16:47.28 | jrun | is AST_CDR_LOCK_APP the same? |
| 16:47.57 | jrun | doesn't look like it though |
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| 18:20.49 | lvlinux | Is there any benefit/difference to using the DTMF feature code to park a call vs having a phone do a SIP refer to transfer the call to 700?? |
| 18:21.06 | lvlinux | (I'm using the default 700 for the parking lot) |
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| 20:30.55 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.9.1 (2016/05/13), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 21:42.03 | tuxd00d | Are there any other ways to hang up a stuck channel (in this case âAutodestruct on dialogâ), other than âchannel request hangupâ or restarting Asterisk? âchannel request hangup SIP/â¦â appears to be ineffective in this case. |
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| 22:51.57 | *** part/#asterisk pckz (~pckz@pc-66-205-83-200.cm.vtr.net) |
| 23:00.39 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 23:05.16 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-85-237.hsd1.or.comcast.net) |
| 23:05.38 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:10.37 | *** join/#asterisk Draecos (~Draecos@120.16.173.187) |
| 23:12.11 | *** join/#asterisk F2Knight (~F2Knight@wsip-64-147-6-99.oc.oc.cox.net) |
| 23:13.00 | *** part/#asterisk kharwell (kharwell@nat/digium/x-ttxkzozhargvcvjs) |
| 23:16.26 | *** join/#asterisk Draecos (~Draecos@120.16.173.187) |
| 23:49.13 | *** join/#asterisk K0HAX_ (~michael@2604:8800:187:1::20) |
| 23:59.55 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |