| 05:51.32 | *** join/#asterisk infobot (ibot@69-58-76-73.ut.vivintwireless.net) |
| 05:51.32 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 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 |
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| 06:21.21 | caesar305 | === --- ---> Locked Here: res_odbc.c line 593 (ast_odbc_direct_execute) |
| 06:21.32 | caesar305 | i see many of these locks when the calls get stuck |
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| 07:19.00 | seik0 | hiagain. ok, prepared solving IAX2 problem =) |
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| 07:36.12 | seik0 | ok, even first suggestion was ok: IAX2 does not work without timing |
| 07:36.59 | seik0 | I needed res_timing_timerfd.so module for iax2 to work fine |
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| 07:56.58 | seik0 | And also I need res_crypto.so, but that is obvious from warnings |
| 08:08.55 | seik0 | though asterisk don't want to build res_crypto for some reason |
| 08:19.09 | seik0 | configure shows "checking for openssl... openssl", but make menuselect does not allow to check res_crypto |
| 08:21.51 | WIMPy | Oh. You didn't have any timing module at all? |
| 08:22.53 | seik0 | There was no need in them |
| 08:23.22 | WIMPy | didn't know Asterisk worked without at all. |
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| 08:42.14 | seik0 | build openssl from sources and still not getting res_crytpo |
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| 08:59.51 | seik0 | explicit installing libopenssl-devel (from rpm) helped |
| 09:00.09 | seik0 | strange, sources always were enough in such cases |
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| 10:12.49 | jaflong | Hi Community. How could a ring tone be dectected on a call? |
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| 10:16.57 | Chainsaw | jaflong: By sending a SIP header, if the phone in question supports it. |
| 10:18.20 | jaflong | I want to on on the actual media level - audio tone |
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| 10:42.04 | mastahh | I am having an odd issue in my Asterisk installation that Asterisk cannot see the res_odbc module at all (even though res_odbc is selected in menuselect). When I "?" in asterisk, the ODBC option is not even there! |
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| 11:01.02 | seik0 | try "module load res_odbc.so" |
| 11:01.06 | seik0 | see if any errors |
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| 13:21.02 | seik0 | You pointed me to jkroon for asterisk multiinstance on one machine, but I can't connect with him. Maybe he's on vacations or smth like that |
| 13:21.26 | seik0 | Someone told I should "query" him, what does it mean? |
| 13:22.10 | WIMPy | Send a private message. |
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| 13:28.10 | seik0 | Server will keep them until he online? |
| 13:28.31 | WIMPy | He is online. |
| 13:29.14 | seik0 | I cannot see him! |
| 13:29.23 | seik0 | Jayk_? |
| 13:29.34 | seik0 | no |
| 13:29.36 | WIMPy | jkroon |
| 13:30.22 | seik0 | I don't know, can't see |
| 13:31.00 | seik0 | anyway, sent msg, but don't know =) |
| 13:32.04 | seik0 | I see 204 online, noone is like jkroon |
| 13:33.17 | WIMPy | There are many thousand more users online. |
| 13:33.49 | seik0 | o.0, so how it is possible? |
| 13:35.02 | WIMPy | man irc :-) |
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| 13:47.45 | mastahh | seik0, I get the error that the module does not exist |
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| 13:48.59 | mastahh | seik0 - I take that back. I made some amendments to the INI files and it seems to have loaded. Let me test further :) |
| 13:49.35 | seik0 | ok, no problem =) |
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| 14:23.17 | seik0 | WIMPy: one channel is not one server, ok =) |
| 14:23.50 | WIMPy | And one server is not one network. |
| 14:25.29 | seik0 | but network limits the universe |
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| 14:35.08 | pic0frame | hey guys! |
| 14:35.57 | pic0frame | what advise can you give on redundancy / failover with Asterisk |
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| 14:48.00 | dorphalsig | Hey Everybody. I asked this q yesterday, but I had to leave before somebody answered. I'm trying to install res_fax_spandsp on an asterisk 13 that was installed via RPM |
| 14:48.10 | dorphalsig | is there an RPM for that module? |
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| 15:12.06 | pic0frame | hey dorphalsig, http://www.rpmfind.net/linux/rpm2html/search.php?query=res_fax_spandsp.so%28%29%2864bit%29 |
| 15:15.43 | pic0frame | http://blogs.digium.com/tag/asterisk-13/ |
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| 17:00.21 | ledoktre | greetings! To anyone out there, Ive got an interesting issue. I was on * 1.8 certified LTS, and just last night made a move over to 13 certified LTS. Everything appeared to compile and install fine - had to download a new g729 codec from Digium, but everything started back up. Problem now appears that when I call in to my box via PSTN etc, I am getting some weird stuttering in the audio. Never had that before with 1.8. Where does a guy start to |
| 17:00.22 | ledoktre | figure that one out? |
| 17:01.54 | ledoktre | literally tested it, worked fine, upgraded to 13, tested - had the stuttering. |
| 17:10.29 | ledoktre | maybe figured it out. changed g729 from barcelona (recommended) to generic. seems to be smooth now. |
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| 17:43.27 | Sanfred | Hi. I'm trying to make two registrations to the same host, but all incoming calls to one of the phone numbers fail to authenticate, "username mismatch". https://issues.asterisk.org/jira/browse/ASTERISK-9395 |
| 17:43.55 | Sanfred | oej has said that there is a lot of documentation. But where? What should I do to solve this? |
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| 17:50.09 | [TK]D-Fender | very few proivders send authenticated calls at all |
| 17:50.19 | [TK]D-Fender | And you are matching them against a peer which matches by IP |
| 17:50.22 | [TK]D-Fender | that has a username in it |
| 17:50.28 | [TK]D-Fender | and it will NOT match 3 possiblities |
| 17:50.45 | [TK]D-Fender | So "type=peer" is an automatci fail on username match for auth |
| 17:51.00 | [TK]D-Fender | So start by removing the challenge your system is making to the inbound call |
| 17:51.06 | [TK]D-Fender | insecure=port,invite |
| 17:51.07 | [TK]D-Fender | ^^^^ |
| 17:57.06 | Sanfred | but the problem is that it matches the wrong peer. "check_auth: username mismatch, have <u0812104026>, digest has <u0812104027>" |
| 17:58.02 | Sanfred | and if I call the other number, it works, without insecure. So I think the provider authenticates, but asterisk can't match it correctly. |
| 17:58.14 | [TK]D-Fender | peer matches by IP <- |
| 17:58.17 | [TK]D-Fender | The host is the SAME |
| 17:58.28 | [TK]D-Fender | It will always match the first no matter what being type=peer |
| 17:58.52 | Sanfred | okay, so what should I use instead of peer? |
| 17:58.58 | [TK]D-Fender | And this still assumes they send you authed calls at all |
| 17:59.03 | [TK]D-Fender | ^ |
| 17:59.16 | [TK]D-Fender | If they aren't sending auth, then even if there's a name there might be no pass |
| 17:59.37 | [TK]D-Fender | If they do full auth (which I've almost never seen), then you'll have to create USER entries for them |
| 18:01.25 | Sanfred | The provider sends "INVITE sip:u0812104027@46.253.192.221:5060 SIP/2.0", isn't there any way to make asterisk read the username part of the invite? |
| 18:07.41 | [TK]D-Fender | <[TK]D-Fender> If they do full auth (which I've almost never seen), then you'll have to create USER entries for them <--- |
| 18:07.49 | [TK]D-Fender | user = match by username |
| 18:08.11 | [TK]D-Fender | Which will still fail if they don't sen full auth with their calls |
| 18:09.59 | Sanfred | Ah, I changed to type=user, and now it works fine. Thank you! |
| 18:10.23 | Sanfred | Obviously my provider is better than most ;) |
| 18:10.34 | [TK]D-Fender | No, now try to call OUT.... |
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| 18:48.54 | linocisco | hi all |
| 18:49.22 | linocisco | I hope everybody has play the audio "Hello World" |
| 18:50.47 | linocisco | Does Asterisk has native feature to play the speech or announcement 2x 3x faster if sysadmin feel that it is lazy voice of Allison Smith |
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| 18:53.37 | [TK]D-Fender | *'s playback apps take what is in the audio file |
| 18:53.48 | [TK]D-Fender | You want it faster... go chnge the recording |
| 18:54.42 | linocisco | [TK]D-Fender, meaning I have to use audacity or any audio editing software to tweak the default asterisk sound files? |
| 18:54.51 | [TK]D-Fender | Audio is what it is |
| 18:54.56 | [TK]D-Fender | edit it yourself |
| 18:55.00 | [TK]D-Fender | there is no "play faster" |
| 18:55.28 | file | doesn't mean it'll sound good, though |
| 18:56.21 | [TK]D-Fender | Why start thinking now? |
| 18:58.58 | linocisco | [TK]D-Fender, thinking to play faster? some prompts seem slower to hear |
| 19:00.14 | DivideBy0 | how's that "3 day weekend" treating you file? |
| 19:01.03 | file | DivideBy0, well! shopping and lunch was good |
| 19:02.02 | [TK]D-Fender | linocisco, Files are files. |
| 19:03.00 | linocisco | [TK]D-Fender, I know file(s). |
| 19:03.30 | [TK]D-Fender | linocisco, They aren't slow. They are what they are. Go edit them if you feel like it |
| 19:04.29 | linocisco | [TK]D-Fender, yes. compared to some prompts from Avaya and Cisco. |
| 19:05.47 | [TK]D-Fender | linocisco, Who says they aren't "fast"? |
| 19:06.15 | [TK]D-Fender | linocisco, perhaps your assumption of what it baseline is faulty |
| 19:06.32 | [TK]D-Fender | linocisco, Either way, you have the files, do whatever you want to them |
| 19:07.02 | linocisco | [TK]D-Fender, I myself feel that. ok let it be if asterisk dosn't have function to play 2x or 3x faster. thanks |
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| 19:24.30 | drmessano | So a couple of my users take too long to explain things |
| 19:24.34 | drmessano | Can Asterisk speed them up? |
| 19:24.42 | drmessano | Is there a SlowUser() function? |
| 19:25.07 | drmessano | or UserSpeed option for Dial()? |
| 19:25.16 | drmessano | Help me now, please, and make it fast |
| 19:30.03 | robmal | You could make a voicemail-like custom dialplan, that would record their channel, occasionaly playing 'uhm' and 'go on' from another, then speed up the recording by 1.5,play it to you and bridge you to the caller. |
| 19:30.37 | robmal | I'd call it 'Gramma Style' |
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| 21:00.11 | dorphalsig | Hello |
| 21:00.41 | dorphalsig | I'm trying to get app_fax_spandsp to work in asterisk13 |
| 21:01.07 | dorphalsig | Sorry |
| 21:01.14 | dorphalsig | res_fax_spandsp |
| 21:01.50 | dorphalsig | I installed asterisk from RPM and then tried to build the .so from source and just copy it |
| 21:02.03 | dorphalsig | asterisk then whined about the compile options |
| 21:02.55 | dorphalsig | Ive looked around for the rpm of that module, but it seems like its only done up to asterisk 11 |
| 21:03.37 | dorphalsig | So I figured I could ask here if anybody know the compile options of the rpm |
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| 21:11.22 | MertsA | you might want to try getting the source rpm from whatever source you installed asterisk 13 from and go from there |
| 21:11.56 | MertsA | unfortunately it looks like the digium repo I use for Centos 6 doesn't have any packages that provide res_fax_spandsp |
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| 21:35.15 | MertsA | So I've got a confusing problem |
| 21:36.20 | MertsA | I have an asterisk 13 install that's pretty standard, if someone calls in to a number that dials a SPA504g cisco phone it rings just fine and everything works including transferring calls |
| 21:37.05 | MertsA | however, if the local cisco phone is set up to forward all calls, asterisk immediately gets a 302 redirect |
| 21:37.35 | MertsA | and it calls whatever number it was forwarded to and adds the incoming and outgoing calls to two bridges |
| 21:37.57 | MertsA | but no audio seems to be passed over the bridge, it's just silence on both sides |
| 21:38.15 | MertsA | it worked in asterisk 11 but after migrating to asterisk 13 it broke |
| 21:38.28 | MertsA | where do I even start to try to troubleshoot this? |
| 21:43.59 | *** join/#asterisk shazaum (~shazaum@unaffiliated/shazaum) |
| 22:03.32 | [TK]D-Fender | look at the actual call. 302 only redirects to the dialplan so the 302 itself isn't a real factor. It's all about where the call came in fron and how it goes back out |
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