| 00:38.25 | ChannelZ-Wk | pee rip? |
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| 00:58.06 | igcewieling | ChannelZ http://www.thinkgeek.com/product/5e09/ |
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| 02:15.22 | j4jackj | Hello? |
| 02:18.13 | hjf | Hello? This is dog |
| 02:19.33 | j4jackj | I'm random. Icecast sucked eggs (and media) through the thinnest straw you will ever see, so I used SSH for audio streaming, and kazam it worked. |
| 02:27.23 | ChannelZ | igcewieling: heh nice |
| 02:29.24 | igcewieling | ChannelZ: took me a while to remember where I saw it, thought it was http://www.tshirthell.com/babyhell.shtml (NSFW) |
| 02:38.12 | j4jackj | I think the ISDN telephone network should move to Opus for better sounding teleconferences. |
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| 02:52.09 | igcewieling | g.722 is already supported on ISDN |
| 02:53.54 | igcewieling | since the b-channels are clear channel you can use pretty much any codec you want on ISDN, as long as both ends of the call agree on the codec |
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| 03:09.42 | WIMPy | Which is a pretty normal thing for conference systems to do. |
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| 04:42.50 | *** join/#asterisk infobot (~infobot@184-229-126-180.pools.spcsdns.net) |
| 04:42.50 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.5.1 (2013/08/27), 10.12.3 (2013/08/27), 1.8.23.1 (2013/08/27), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 06:15.05 | ChannelZ | mucho |
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| 10:44.42 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.5.1 (2013/08/27), 10.12.3 (2013/08/27), 1.8.23.1 (2013/08/27), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 11:56.23 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.5.1 (2013/08/27), 10.12.3 (2013/08/27), 1.8.23.1 (2013/08/27), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 18:52.32 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.5.1 (2013/08/27), 10.12.3 (2013/08/27), 1.8.23.1 (2013/08/27), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 18:55.28 | [TK]D-Fender | itsplist-us |
| 18:55.34 | [TK]D-Fender | ~itsplist-us |
| 18:55.35 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
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| 19:14.16 | i_am_good | Can someone take a look at my look, please? https://gist.github.com/anonymous/6478384 |
| 19:14.31 | i_am_good | I think I made a mistake in my extensions, but can't catch it |
| 19:16.13 | [TK]D-Fender | A look at your look? |
| 19:16.15 | i_am_good | STUPID ME |
| 19:16.25 | i_am_good | forgot to comment out |
| 19:16.29 | [TK]D-Fender | All I see you is tossing your calls at "Congestion" |
| 19:16.38 | [TK]D-Fender | Which is what that CDR is showing too |
| 19:16.57 | [TK]D-Fender | "","2066820185","2532433035","default","""unknown"" <2066820185>","SIP/66.54.140.46-00000004","","Congestion","","2013-09-07 20:11:54",,"2013-09-07 20:11:54",0,0,"NO ANSWER","DOCUMENTATION","1378581114.4","" |
| 19:17.03 | [TK]D-Fender | exten => _X.,1,Congestion |
| 19:17.20 | eirirs | hahaha |
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| 21:14.51 | Alex25 | Every time compiled asterisk-1.8.3.3 from source on a Debian7, chan_sip is not available |
| 21:14.54 | Alex25 | it is shown as "XXX chan_sip" on menuselect screen, means not available |
| 21:15.14 | Alex25 | I had the same problem on a different Debian7 VPS, so it looks like some conflict on this distribution |
| 21:15.27 | Alex25 | Where should I look for a solution?? |
| 21:18.22 | ChannelZ | Missing a prereq I guess. Is res_crypto selectable under Resource Modules? |
| 21:18.55 | [TK]D-Fender | menuselect says what you're missing |
| 21:19.04 | ChannelZ | (and are you needing 1.8 for a specific reason? Why not build Asterisk 11 if you're going to the trouble?) |
| 21:20.02 | Alex25 | no, res_crypto also XXX\ |
| 21:20.10 | ChannelZ | so you need openssl-dev |
| 21:20.58 | ChannelZ | or just openssl, there might not be a -dev .. |
| 21:21.38 | Alex25 | thanks, let me check on apt |
| 21:22.16 | ChannelZ | libssl-dev |
| 21:22.34 | Alex25 | any other component could be missing? just to be sure before i recompile again.. |
| 21:23.01 | igcewieling | I imagine a google search would tell you what libraries you need |
| 21:23.11 | ChannelZ | libxml2-dev |
| 21:23.17 | ChannelZ | but RE: why are you building 1.8 |
| 21:24.58 | Alex25 | I need 1.8 because I;m used to it, and last time I tried to migrate there was a difference with the way some applications worked |
| 21:25.54 | Alex25 | I had this version on Debian 6 it worked fine |
| 21:27.52 | ChannelZ | Yeah it works, just curious. I never had a hard time migrating. It is LTS I guess |
| 21:30.40 | Alex25 | so that seems not to be the problem, I already have openssl and libssl-dev installed.. |
| 21:30.55 | Alex25 | any other hint? |
| 21:31.18 | ChannelZ | well I don't know about 1.8 specifically.. what does it say for the requirements when you highlight the module in menuconfig? |
| 21:32.24 | igcewieling | Alextry http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-Install |
| 21:32.29 | ChannelZ | in 11 it wants chan_local which has no deps of its own so that should already be on, and res_http_websocket which also has no deps of its own. |
| 21:32.50 | Alex25 | it says "Depends on: chan_local(M), res_crypto(M)" |
| 21:33.29 | igcewieling | did you re-run ./configure after you installed the -dev packages? |
| 21:33.32 | ChannelZ | well res_crypto not enabled is blocking that, but not sure what's blocking res_crypto if you say you already have libssl-dev |
| 21:34.04 | ChannelZ | Yeah was gonna say reconfig but he said "I already have libssl-dev installed" |
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| 21:35.41 | Alex25 | is seems like specific problem on Debian7, I had the same issue on another VPS |
| 21:36.28 | igcewieling | Alex25: Open a bug on Jira |
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| 21:38.24 | ChannelZ | libopenssl-devel maybe? |
| 21:38.38 | igcewieling | You'd think if it is a problem specific to Debian 7, then thousands of people would be complaining about it. |
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| 22:12.51 | Freeaqingme | If a binary codec works under asterisk 11, can it be assumed to be working under 12 as well? |
| 22:23.35 | slav3_kitten | you know what they say about assumptions |
| 22:26.52 | igcewieling | what exactly is a binary codec? |
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| 22:30.38 | i_am_good | Hi, I am testing my asterisk with an IPKall number, but when I check my calls, but it's not passing the caller ID |
| 22:30.55 | Freeaqingme | igcewieling, I meant a precompiled version, as they're distributed by Digium, like G729 and SILK |
| 22:31.26 | igcewieling | Freeaqingme: they are unlikely to work between major versions. |
| 22:32.26 | i_am_good | Here's my extensions.conf and CDR log: https://gist.github.com/anonymous/6479986 |
| 22:37.19 | ChannelZ | I don't actually see you calling anything |
| 22:38.15 | i_am_good | how can I better show you my log? |
| 22:38.16 | ChannelZ | Freeaqingme: they will build new binaries by release |
| 22:38.24 | i_am_good | oh |
| 22:38.26 | i_am_good | sorry :D |
| 22:38.34 | ChannelZ | i_am_good: verbose console output would be a good start |
| 22:39.08 | ChannelZ | and what do you mean by "when I check my calls it's not passing the caller ID" - are you talking about an incoming call going to a local device? |
| 22:39.15 | Freeaqingme | ChannelZ, I was assuming that. I'm setting up a new server now, so would be handy if they were now available.That's it ;) |
| 22:40.05 | ChannelZ | ah yeah you might be screwed on that for a little while. |
| 22:40.25 | i_am_good | ChannelZ: https://gist.github.com/anonymous/342eb38389159be72573 |
| 22:40.44 | i_am_good | Yes, my DID is not passing the actual caller's phone number |
| 22:40.55 | Freeaqingme | ChannelZ, I'm a patient man. Also, contacting Digium, see if they already have a QA build available I could borrow |
| 22:41.43 | ChannelZ | i_am_good: do a NoOp(${CALLERID(all)}) and see if you're even getting it |
| 22:43.55 | i_am_good | ChannelZ: https://gist.github.com/anonymous/8e7539a4598d0f483e56 |
| 22:44.01 | i_am_good | I don't think it's even passing it |
| 22:44.07 | ChannelZ | and actually judging by your CDR you are getting it, calling from 2066820185 |
| 22:44.30 | i_am_good | No, I am calling from a 347..... number |
| 22:44.42 | i_am_good | I think IPkall is sending wrong CID |
| 22:44.48 | ChannelZ | Well you're being sent 2066820185, whatever that is. |
| 22:45.05 | ChannelZ | If that's not right then that's your ITSP |
| 22:45.29 | i_am_good | Got it. Thought I screwed up something in my extensions. Thanks for the help! |
| 22:45.46 | ChannelZ | what is the 206 number? Does it have any relevance at all? |
| 22:46.38 | ChannelZ | http://800notes.com/Phone.aspx/1-206-682-0185 |
| 22:47.07 | ChannelZ | interesting historuy |
| 22:47.09 | i_am_good | It's a free number from ipkall.com |
| 22:47.34 | i_am_good | I think they make money from traffic or something |
| 22:47.44 | ChannelZ | but not the number you're actually calling? |
| 22:48.13 | i_am_good | No, I was assigned number ending in ...3035 but it's coming as ...0185 |
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| 22:48.41 | i_am_good | Looks like it's a common issue with them. Well, it's ok for testing |
| 22:49.02 | ChannelZ | Got what you paid for I guess |
| 22:49.11 | i_am_good | Exactly :D |
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