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| 03:20.08 | UncleKiwi | hi, just wondering if the cdr can out put call time in HH:MM:SS |
| 03:20.10 | UncleKiwi | ? |
| 03:20.37 | UncleKiwi | sorry call duration |
| 03:46.01 | [TK]D-Fender | CDR doesn't pull anything |
| 03:46.17 | [TK]D-Fender | misread that |
| 03:47.06 | [TK]D-Fender | Maybe you can use a date function in a custom CDR storage definition |
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| 09:33.15 | woopstar | Hi. Is there any great wiki/tutorial/documentation on how to interconnect two asterisk servers (version 13 please). We have one in US and on in EU. We'd like to hook them up, so users can register there phone to each server and call each other. |
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| 10:03.19 | MaliutaLap | woopstar: http://wiki.freepbx.org/pages/viewpage.action?pageId=4161588 http://asteriskguide.quora.com/Create-a-SIP-trunk-between-two-Asterisk-Servers http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-OutsideConn.html |
| 10:03.44 | woopstar | Thank you very much |
| 10:04.13 | woopstar | would you prefer IAX2 or SIP for connecting two servers? |
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| 10:43.15 | eirirs | IAX2 |
| 10:43.32 | eirirs | it's what IAX2 are for. |
| 10:44.08 | eirirs | IAX2 are short for Inter-Asterisk eXchange Version 2 - https://tools.ietf.org/html/rfc5456 |
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| 12:05.16 | mugove | I am trying to use AMI to put a call on hold |
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| 12:05.48 | mugove | how do I do it |
| 12:09.23 | WIMPy | You can not put calls on hold from the server side. It needs to be done on the terminal. |
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| 12:10.11 | WIMPy | What are you trying to do? |
| 12:10.22 | mugove | ok. For example(AMI)? |
| 12:11.23 | Greenlight | I think there's a Park action in the AMI |
| 12:14.39 | WIMPy | Or you can redirect the call wherever. But those are different things. |
| 12:15.37 | Greenlight | Indeed. We have a "hold" extnesion that we Redirect to using the AMI so that users can place calls on "hold" that way |
| 12:16.18 | WIMPy | Just that the call wouldn't be on hold. For neither end. |
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| 12:20.16 | woopstar | Hi. ODBC is working fine, connection is running. iaxfriends table is already present from the alembic setup. Added "iaxpeers => odbc,asterisk_write,iaxfriends" to extconfig. But when adding a peer, it does come by when doing a show peers on iax2. What could be wrong? |
| 12:23.08 | woopstar | It does say "== Binding iaxpeers to odbc/asterisk_write/iaxfriends" when asterisk is starting |
| 12:23.09 | RoyK | Since it doesn't look like there's anything like an FXS adaptor usable for a rapsberry pi out there, I just wonder - does anyone know how hard it could be to make one? I know a bunch of people very knowledgable in electronics, but I have no idea where to find the docs on how POTS really works |
| 12:24.20 | WIMPy | RoyK: There should be some USB ones. And otherwise there's a lot of ATAs to choose from. |
| 12:25.20 | RoyK | WIMPy: ATAs implies IP telephony and what I want to make, is an FXS-to-bluetooth that also support pulse dialling |
| 12:26.08 | WIMPy | Im sure there should be ATAs that support pulse dialling. |
| 12:26.12 | RoyK | WIMPy: and there "should be", yes, but all I can find is things in the area of $250 |
| 12:27.05 | RoyK | WIMPy: yes, but as I said, ATAs impy IP telephony, and they take up room, won't fit into an old telephone :P |
| 12:27.21 | RoyK | I'm talking about chan_alsa <-> chan_phone |
| 12:27.27 | WIMPy | Yea, for whatever reason it seems to be a lot cheaper to take an additional step via ISDN. |
| 12:27.59 | WIMPy | But I'd recommend just using an ATA. Everything else will end in headaches. |
| 12:28.10 | RoyK | and how would you do that with a raspberry pi? you'd still need the ISDN interface and some adaptors |
| 12:28.54 | WIMPy | For the console channels you don't need more than a sound"card". But you still have to find a way to dial. |
| 12:29.23 | WIMPy | Yes, but USB ISDN adapters are cheap. |
| 12:29.48 | RoyK | and FXS to ISDN? |
| 12:30.06 | WIMPy | However I didn't manage to compile LCR on the Pi, at least not on the 1st attempt. |
| 12:30.35 | WIMPy | Available in big numbers in charity sales. |
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| 12:30.59 | WIMPy | Or ebay off course. |
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| 13:25.04 | gustopn | LOL |
| 13:25.17 | gustopn | there are still 1.8 comming out? |
| 13:26.40 | gustopn | WIMPy, are you there? |
| 13:26.45 | gustopn | Penguin, hi |
| 13:26.46 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 13:26.47 | [TK]D-Fender | ^^^ |
| 13:27.03 | [TK]D-Fender | 1.8 dies in 5 months |
| 13:27.13 | gustopn | yes |
| 13:27.26 | gustopn | that's why I wonder that there are still comming new releases |
| 13:27.36 | gustopn | 1.8.32.3 (2015/04/08) |
| 13:27.57 | [TK]D-Fender | Because it dies in 5 months |
| 13:28.02 | gustopn | yes |
| 13:28.07 | gustopn | that I understood |
| 13:28.14 | [TK]D-Fender | Then you shouldn't wonder |
| 13:28.43 | gustopn | however, asterisk grew quite fast last years |
| 13:29.38 | gustopn | I am moving to 13 with everything except the openwrt ones |
| 13:30.28 | gustopn | and am happy that my new phone works now with g722, so I enjoy better sound quality |
| 13:31.42 | gustopn | the next move will be to try if it can pick the right codec according to the phone that is called (either g722 for the new one or g711 for the old one) |
| 13:32.23 | gustopn | I noticed that in recent versions it does not only look at what is configured on the peer that is the first in line, but backwards, that is good, but broke some of my old configurations back then |
| 13:32.40 | gustopn | now I am counting on that behaviour, otherwise someone will have to transcode |
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| 14:17.14 | Greenlight | Hmm does 13 no longer create the asterisk user and groups during install ? |
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| 14:20.50 | file | Asterisk has never done such things |
| 14:20.52 | mjordan | I'm not aware of it ever creating a predefined user/group |
| 14:21.06 | Greenlight | Oh, I was sure 11 and earlier had created them itself |
| 14:22.34 | Greenlight | Perhaps it's just been a while I've installed fresh instead of upgrading, and I've just forgot :) |
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| 15:41.47 | Greenlight | On the CLI in 13, does anyone else have the issue where the color persists onto the current CLI entry? |
| 15:42.03 | Greenlight | Like, if the last entry was in purple,the cli entry is now in purple |
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| 15:50.13 | overyander | can someone tell me what is wrong with this call file? here's the file and the error asterisk is giving me. https://gist.github.com/jeffclay/87d8a161fe5a3d08f9df |
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| 15:54.41 | [TK]D-Fender | overyander: show us how you're putting it there.... |
| 15:59.31 | overyander | [TK]D-Fender, I'm using a python script that generates the file localy on another system and then uses scp to put the call file in the outgoing dir. |
| 16:00.05 | overyander | scp.put(callfile, '/var/spool/asterisk/outgoing/' + callfile) |
| 16:00.24 | [TK]D-Fender | that is not legit |
| 16:00.35 | [TK]D-Fender | * will lock that file INSTANTLY as soon as it's there |
| 16:00.40 | [TK]D-Fender | And read it INCOMPLETE |
| 16:01.02 | [TK]D-Fender | You are required to MOVE the file into the spool folder from the same filesystem |
| 16:01.19 | overyander | oh |
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| 16:18.32 | overyander | [TK]D-Fender, even when moving with this command locally, it fails with same error "mv /var/spool/asterisk/tmp/14565552930-fax-noreply-2697.call /var/spool/asterisk/outgoing/" |
| 16:19.50 | [TK]D-Fender | show us |
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| 16:20.35 | overyander | show you what? the error? |
| 16:20.50 | [TK]D-Fender | yes, and what you can for your new code |
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| 16:47.19 | tuc0 | hey guys, does anyone have a recommendation for the best way to do voice to text? commercial options included? |
| 16:49.16 | [TK]D-Fender | All STT is inherently crap. Over any telephony standard it's significantly worse |
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| 17:06.40 | tuc0 | I don't know Google Translate as well as Amazon's system seems to work pretty well. I had a bot call me not too long ago that was pretty convincing until I asked it to add 1+2 |
| 17:07.32 | tuc0 | Then it creepily laughed at the question. Part of me wondered if it was someone with a thick accent behind a sound board. |
| 17:08.56 | tuc0 | [TK]D-Fender, also, I don't need perfect sentence reconstruction more like "yes/no" "0-9" kind of things |
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| 17:38.06 | [TK]D-Fender | tuc0: then Sphinx / Lumenvox, etc |
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| 17:52.46 | tuc0 | [TK]D-Fender, thank you |
| 18:12.04 | overyander | i can receive faxes via t.38 fine, i have the free fax for asterisk module installed, yet when i try to send a fax asterisk won't offer t.38 on the channel |
| 18:12.12 | overyander | what am i missing? |
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| 18:21.03 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.3.2 (2015/04/08), 11.17.1 (2015/04/08), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 18:35.21 | overyander | Hey [TK]D-Fender, just found the answer |
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| 18:36.00 | overyander | I tested this and it works... http://www.digium.com/sites/digium/files/fax-for-asterisk-manual.pdf section 3.1.2 says that you can use the 'z' option to initiate a t.38 re-invite. |
| 18:37.22 | [TK]D-Fender | interesting. |
| 18:37.29 | [TK]D-Fender | I'd certainly follow that if it says it... |
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| 19:21.40 | fling | channel.c:5211 in set_format: Unable to find a codec translation path from 0x1 (g723) to 0xf14e (gsm|ulaw|alaw|slin|g729|g722|slin16|siren7|siren14) |
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| 19:22.49 | fling | Which codec from g711/723/729 has the better sound quality? |
| 19:23.21 | WIMPy | 711 |
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| 19:26.25 | [TK]D-Fender | * can't legally transcode G.723 except for expensive transcoder cards. |
| 19:28.57 | RoyK | [TK]D-Fender: didn't it have a codec for that? |
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| 19:29.42 | [TK]D-Fender | No |
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| 19:42.58 | fling | Ok, thanks for the info. |
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| 20:02.15 | fling | Why is not this working?: |
| 20:02.27 | fling | exten => 0123,1,GoTo(skype&echo123) |
| 20:02.30 | fling | exten => _skype&.,1,Set(ADDID=${EXTEN}) ⦠|
| 20:03.18 | fling | ohh am I fogrot ,1 ? ok |
| 20:03.29 | *** join/#asterisk infina (~infina@unaffiliated/infina) |
| 20:05.11 | fling | ok, it works |
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| 20:46.39 | deltalima | oops |
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| 21:23.18 | *** mode/#asterisk [+o file] by ChanServ |
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| 22:05.45 | iulhk | using asterisk 11 , at the cli how can i know the connected call is audio or video ? |
| 22:07.37 | iulhk | if 2 sip peeers connected, when peerA switching Audio to Video or Video to Audio, do client send any re-invite or can i capture this re-invite ? |
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| 22:08.26 | Greenlight | Asterisk is at the core of a lot of our business. So I was wondering; the Digium Support packages - what sorts of issues are they able to assist with? For example if we're experiencing deadlocks, would having a Digium Support package help get the issue looked at quicker? |
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