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| 01:03.30 | volga629 | Hello Everyone, how to convert properly audio file for asterisk sox main_IVR_greeting.wav -t raw -r 8000 -c 1 main_IVR_greeting.sln |
| 01:03.50 | volga629 | this example didn't worked |
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| 02:27.05 | WIMPy | Have you tried to ask Dr. Google for Asterisc and sox? It should come up with tons of examples. |
| 02:40.25 | volga629 | yes I tried sox and ffpmeg |
| 02:44.25 | WIMPy | I don't see a flag there for signed linear 16 bits. And the basic format is usually a .wav file. |
| 02:52.22 | volga629 | for output file correct ? |
| 02:53.51 | WIMPy | sure |
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| 03:36.31 | volga629 | for destination like this -r 16000 -b -c 2 -t wav ? |
| 03:38.46 | WIMPy | That won't work. |
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| 03:58.36 | volga629 | I think too tried nothing |
| 04:04.08 | volga629 | @rbuz4758 |
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| 04:14.30 | ChannelZ | Can someone clarify my brain with RTCP stats; rlp is remote lost packets, but that's from the remote end's point of view right? So that's packets the remote end lost from us? Or am I backwards? |
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| 04:34.27 | ChannelZ | brb |
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| 04:38.33 | volga629 | what is correct options ? |
| 04:46.23 | ChannelZ | for |
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| 09:41.17 | pawiecki | good morning! I want to add new account in a "crowded" sip.conf - how can i quickly find number that's not already in use? |
| 09:42.36 | pawiecki | hmm, i think i could grep it somehow |
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| 10:31.36 | catphish | using queues, is it possible for an agent to reject a specific call and not have it call their channel again? but still receive all future calls? |
| 10:34.50 | catphish | this would be simple with a round-robin strategy, they could reject the call and it would move on, but with with ringall, the call immediately comes back after being rejected |
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| 10:45.42 | TheEndIsNear | does anyone have any experience with creating an asterisk service for firewalld? |
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| 11:29.44 | elfranne | i just installed the newly released 13.1.0 but when i starts asterisk the /var/run/asterisk/asterisk.ctl and pid are created but the process is not present is the process list. No error present in the log apart from some sounds missing |
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| 12:59.49 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.1.0 (2014/12/15), 11.15.0 (2014/12/15), 1.8.32.1 (2014/11/20); Standard: 12.8.0 (2014/12/15); 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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| 13:32.57 | stefan27 | Must a SDP Answer not include codecs that the Offerer did not offer? i.e. if INVITE has m=audio 21724 RTP/AVP 8 0 18 101 then is it wrong to respond with m=audio 12420 RTP/AVP 8 0 9 101? |
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| 14:04.06 | hrnt | anybody familiar with polycom VVX 500 and G.722.1C? I updated its firmware and now suddenly it won't talk siren14 with the PBX. |
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| 14:32.52 | ovoshlook | Hello. Need help with Asterisk 12.6 and cdr_adaptive_odbc. I configured connector, database and cdr_adaptive_odbc.conf correctly. When I do Core reload I see that parser of this file sucessfully reads info from it, but I see no one row at dump wwen reload core. |
| 14:33.31 | ovoshlook | At same machine with 12.7 that connexted with this that database and table all works correctly |
| 14:35.14 | WIMPy | Looks like you already found the solution. |
| 14:36.38 | ovoshlook | ?? It is a bug of 12.6? |
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| 14:37.14 | WIMPy | Did yu check the changelog? |
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| 14:40.03 | ovoshlook | Emm... No. |
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| 14:43.13 | ovoshlook | Now checked. No entries for cdr_adaptive_odbc |
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| 15:53.41 | anthonym | Hey guys, I've setup an asterisk box (using the iso), recorded some welcome greetings in Audacity, using 8000hz sampling, mono, 32 bit PCM (also tried 8 and 16), it sounds awesome on my computer, but when I upload it to the box, and call the number, the recording sounds bloody horrible. |
| 15:53.45 | anthonym | what am i doing wrong? |
| 15:54.01 | anthonym | Ive tried https://snowulf.com/2012/02/21/recording-high-quality-voice-prompts-for-asterisk/ |
| 15:54.06 | leffelad | GSM Wav? |
| 15:54.13 | anthonym | and http://wiki.kolmisoft.com/index.php/Convert_WAV_file_to_Asterisk_playable_format |
| 15:54.23 | anthonym | I saved it as ulaw, I also tried standard 16bit WAV |
| 15:54.28 | WIMPy | You need to make them 16 bit. |
| 15:55.12 | [TK]D-Fender | Also you have not been clear on precisely how that call arrives at your server and its characteristics. that leg of the trip coudl itself be at fault |
| 15:55.14 | WIMPy | And mono, off course. |
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| 15:56.36 | anthonym | thank you. I've tried all that too. mono, 16bit, 8000hz sampling etc. I might have to get some sleep and try again soon |
| 15:59.25 | anthonym | actually, I just did the 16bit wav.. It sounds better, but its like it isnt loud enough |
| 15:59.28 | anthonym | no where near loud enough |
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| 15:59.47 | anthonym | however listening to it on the computer its extremely good quality and loud |
| 15:59.58 | leffelad | pop a bit of amplification on it in audacity |
| 16:00.12 | WIMPy | I usually have the issue that it gets too loud. |
| 16:00.25 | [TK]D-Fender | It's loud in audacity... because you are listening through computer speakers amplified. |
| 16:00.46 | [TK]D-Fender | this is not relative to the volume of a normal call as an audio file |
| 16:01.55 | anthonym | oh SHIT yeah, that worked. Thank you SO SO SO SO much. |
| 16:02.40 | WIMPy | Still better to adjust the volume when recording. |
| 16:05.00 | anthonym | i will make sure i do for the next one :) |
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| 16:42.25 | gaps | using Asterisk 11.2.1, and having queue with rrmemory and joinempty=yes, when there are volume of calls the device state is showing as Ring+Inuse and for the same agent he is getting one more call when he is on the call |
| 16:42.30 | gaps | how to avoid this |
| 16:44.24 | [TK]D-Fender | gaps: Show us |
| 16:44.41 | [TK]D-Fender | gaps: After UPGRADING first. You are significantly behind. |
| 16:45.11 | gaps | my log is here - http://pastebin.com/Dchg9JVj |
| 16:46.07 | gaps | line number 16 |
| 16:46.43 | [TK]D-Fender | 14 is where it tried to call... and 16 it changed to that state (presumably from the attempt to call it). |
| 16:47.03 | [TK]D-Fender | That does not look like An attempt to use it AFTER it has already reached that state |
| 16:47.26 | [TK]D-Fender | Upgrade. The replicate and show a call with just basic verbose 10, and a dump of the que prior, and during |
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| 16:48.05 | gaps | [TK]D-Fender: in line 7 it got answered |
| 16:48.23 | gaps | [TK]D-Fender: but in line 8 still it show as Ring+Inuse |
| 16:48.44 | [TK]D-Fender | gaps: Show a new call with proper dumps. |
| 16:48.54 | gaps | ok just a sec |
| 16:49.17 | [TK]D-Fender | gaps: and UPGRADE. We don't know if your issue was actually real, and possibly solved in the 12 releases since the one you're running. |
| 16:50.29 | gaps | [TK]D-Fender: since this is a production server.. so upgarding is difficult at the point of time... |
| 16:52.39 | [TK]D-Fender | gaps: Provide the debug I request for the version you haave at least |
| 16:53.12 | gaps | [TK]D-Fender: ok let me pull up a call's log |
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| 17:05.55 | [TK]D-Fender | gaps: I want NEW output giving what pieces I asked for and how. |
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| 17:24.49 | gaps | [TK]D-Fender: call volume is high... so at this moment I can't take log for one call... |
| 17:25.02 | gaps | [TK]D-Fender: around 50 calls are going..sorry |
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| 17:49.33 | WIMPy | My headset is starting to fall apart :-( |
| 17:51.12 | volga629_ | Hello Everyone, WIMPy what will be correct way use arguments for sox ? |
| 17:52.21 | volga629_ | still can't convert to proper format |
| 17:55.07 | WIMPy | http://www.google.de/search?q=asterisk+sox |
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| 17:59.01 | volga629_ | I tried bunch example, but didn't worked |
| 17:59.21 | volga629_ | : WAVE: RIFF header not found |
| 18:00.41 | WIMPy | Then maybe you should read again. |
| 18:01.19 | WIMPy | If you save it as .wav it needs a riff/wave header. If you write raw, call it .sln. |
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| 18:13.07 | rrittgarn | so i know realtime isn't heavily leveraged (at least from what i can tell sitting in this channel). I think i have found either a configuration issue with MySQL or a bug within asterisk realtime when using ODBC VM Storage. |
| 18:14.05 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 18:14.53 | rrittgarn | i know this sounds silly, but you can't leave more than 11 VMs at the same time in ideal conditions if one or more of those messages goes over 3 minutes in length. Now aside from the obvious no-no of storing a raw wav in a mysql DB, if this truly is a bug, what's my next step? I've replicated on a few boxes now |
| 18:16.18 | rrittgarn | overall, i think the single threaded nature of the ODBC connetion (using res_odbc) has issues keeping up... |
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| 19:04.53 | marceloamorim | guys, simple question, for set group for each dahdi ( fxo ) channel |
| 19:05.27 | marceloamorim | should I use context=X, channel => 1, group=1 for example? |
| 19:05.53 | marceloamorim | and context=Y, channel => 2, group=1 |
| 19:06.04 | [TK]D-Fender | no. |
| 19:06.09 | WIMPy | All options apply to the FOLLOWING channel=> line. |
| 19:06.27 | [TK]D-Fender | parameters you want applied to channels must be set BEFORE the Channel => directive |
| 19:06.54 | marceloamorim | ohh, so just put the group before the channel |
| 19:06.55 | marceloamorim | nice |
| 19:06.58 | marceloamorim | thx [TK]D-Fender |
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| 19:39.30 | ipengineer | Does âApplication_Authenticateâ no longer exist in asterisk 13? |
| 19:39.36 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:40.00 | [TK]D-Fender | It does |
| 19:40.09 | ipengineer | [TK]D-Fender: Ok thanks |
| 19:40.16 | [TK]D-Fender | Why would you think otherwise? |
| 19:40.43 | ipengineer | When I tried it in the dialplan it crashed asterisk on dialplan reload and I didnt see any documentation for it.. I prob overlooked it. I will go back and dig again |
| 19:40.50 | Gazzonyx | Afternoon *, I'm using the Asterisk chef cookbook. When I provision a machine, I get a base build which is set to listen on all interfaces via TCP and UDP port 5060. When I start Asterisk, the service is only listening on the mgmt port. Any ideas why the SIP port is not even opened? |
| 19:41.22 | [TK]D-Fender | Gazzonyx: You may not have even loaded the module |
| 19:41.37 | [TK]D-Fender | Gazzonyx: Or you may have broken its config causing it to fail to load |
| 19:41.49 | [TK]D-Fender | Gazzonyx: Or you may be looking for it the wrong way and it's there |
| 19:42.32 | Gazzonyx | [TK]D-Fender: Actually, I don't have a modules.conf. The cookbook never created one. I almost guarantee that's the issue. :) |
| 19:43.18 | [TK]D-Fender | I've never seen any book create a file. Many have instructions about this... |
| 19:44.11 | [TK]D-Fender | What book is this exactly? |
| 19:44.15 | [TK]D-Fender | (link please) |
| 19:50.09 | Gazzonyx | [TK]D-Fender: You mean the chef cookbook? |
| 19:50.15 | [TK]D-Fender | yes |
| 19:50.56 | [TK]D-Fender | If you're referring to a physical book in-hand then the ISBN or UPC will do... |
| 19:51.01 | Gazzonyx | https://supermarket.chef.io/cookbooks/asterisk |
| 19:51.48 | Gazzonyx | [TK]D-Fender: No, no, no. I'm talking about a Chef cookbook. It's used for automating install and configuration of servers. :) |
| 19:52.32 | [TK]D-Fender | So I see. Anything third-party like that is a YMMV situation |
| 19:52.43 | [TK]D-Fender | And * provides packages for CentOS |
| 19:58.30 | mjordan | yeah... that cookbook is a bit suspect. |
| 19:58.36 | mjordan | node['asterisk']['sip']['useragent'] - (default 'Asterisk with Adhearsion') |
| 19:58.58 | mjordan | not saying it's a bad starting point, but you'd definitely want to go in and modify it |
| 19:59.22 | mjordan | ah |
| 19:59.25 | mjordan | it was done by ben |
| 19:59.32 | mjordan | that makes sense about the user agent then :-) |
| 20:00.34 | mjordan | Gazzonyx: I can almost guarantee that the author of that cookbook never intended it to be used without modification and/or tweaking for your own deployment. |
| 20:00.48 | mjordan | in fact, it even says that in the preamble |
| 20:00.49 | mjordan | "It is intended that this cookbook remain small and perform only installation tasks, with downstream cookbooks depending on it to configure Asterisk for more specific tasks." |
| 20:03.30 | pabelanger | Gazzonyx, what we do with puppet, is provision a base setup of asterisk. Then, using [context](+) override each setting we need |
| 20:03.37 | pabelanger | so |
| 20:03.41 | pabelanger | by default with sip.conf |
| 20:03.44 | pabelanger | we set |
| 20:03.48 | pabelanger | [general] |
| 20:03.54 | Gazzonyx | mjordan: Haha, yeah. I'm getting that. :) |
| 20:03.55 | pabelanger | bindaddr = 127.0.0.1:5060 |
| 20:04.03 | pabelanger | then we override it as needed |
| 20:04.04 | pabelanger | using |
| 20:04.12 | pabelanger | sip-custom.conf |
| 20:04.13 | *** join/#asterisk ibercom (5337900f@gateway/web/freenode/ip.83.55.144.15) |
| 20:04.17 | pabelanger | [general](+) |
| 20:04.25 | pabelanger | bindaddr = 1.1.1.1:5060 |
| 20:05.36 | [TK]D-Fender | pabelanger: Last I was away you had to use bindaddr and bindport as their traditional separate fields. Has this changed? |
| 20:05.41 | [TK]D-Fender | aware* |
| 20:06.02 | Gazzonyx | pabelanger: That makes sense. We'll probably end up doing something like that. Right now I'm just mocking up a proof of concept that uses asterisk. |
| 20:11.01 | pabelanger | [TK]D-Fender, sorry, udpbindaddr |
| 20:11.11 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 20:11.14 | pabelanger | Gazzonyx, checkout https://github.com/kickstandproject/asterisk/tree/master/debian/ast_config |
| 20:11.19 | pabelanger | for out base configs |
| 20:11.44 | pabelanger | it has been working well for the past 3 years |
| 20:12.31 | pabelanger | and an example template file: https://github.com/kickstandproject/kickstandproject-asterisk/blob/master/templates/etc/asterisk/sip.conf.d/10general.conf.erb |
| 20:12.53 | ibercom | Which version of asterisk consumes more CPU ? A11 or A1.8 |
| 20:13.09 | pabelanger | ibercom, loaded question. What are you doing? |
| 20:13.21 | Gazzonyx | pabelanger: Nice. Unfortunately our infrastructure is tied to Chef ATM. |
| 20:13.36 | ibercom | sip media no transcoding iax and pri |
| 20:13.45 | pabelanger | Gazzonyx, Chef is fine, if you are doing automation and asterisk you are on the right path |
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| 20:14.21 | pabelanger | ibercom, they should consume about the same. |
| 20:14.23 | Gazzonyx | pabelanger: Yeah, we're pretty much a Chef/RHEL shop. |
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| 20:17.39 | ibercom | pabelanger, I thought so. But my A11 consumes twice that A1.8 ??? |
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| 20:19.17 | ibercom | I use the same config and dialplan. |
| 20:19.49 | mjordan | so if you knew the answer to your question, why did you ask it? |
| 20:19.54 | pabelanger | ibercom, hard to said. You'd have to profile asterisk and see what is going on. |
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| 20:21.13 | skrusty | evening |
| 20:22.09 | ibercom | mjordan, I need other experiences. Mine isn't good. |
| 20:25.18 | ibercom | mjordan, A11 consumes twice more CPU and alot more memory. |
| 20:26.06 | [TK]D-Fender | ibercom: With the exact same modules & usage? |
| 20:26.20 | mjordan | what version of Asterisk 11? |
| 20:27.02 | ibercom | Yes, the same config and similar modules. Usage = |
| 20:27.29 | ibercom | Asterisk 11.14. |
| 20:28.21 | ibercom | This is a production server. |
| 20:28.54 | [TK]D-Fender | 11.15 is also out, and you haven't mentioned the 1.8 exact version |
| 20:29.53 | ibercom | The latest versions of asterisk 1.8: 1.8.30 or 1.8.32 |
| 20:30.18 | mjordan | well, the last performance improvement that went in was long before that, so I don't think upgrading to 11.15 would change much. There was a minor improvement to switch performance in 11.15. |
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| 20:30.40 | mjordan | But basically, what pabelanger said. You'll need to profile it to determine why it is running slowly. No one here can guess what your system is doing. |
| 20:31.34 | mjordan | hold on. |
| 20:31.53 | ibercom | I expected the performance was similar. |
| 20:31.55 | mjordan | you filed ASTERISK-24608. Barring the memory aspects |
| 20:31.59 | mjordan | ibercom: you would be wrong |
| 20:32.10 | mjordan | you're complaining about a jump from 1.8% to 5%? |
| 20:32.33 | mjordan | slow memory growth, yes, that's bad |
| 20:33.23 | mjordan | this - https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats - required changes to the media format core. Since we no longer could use a bit field for determining media types, there's a bit more overhead in some media processing |
| 20:33.33 | mjordan | there is no free lunch in software. |
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| 20:34.30 | ibercom | Yes but I don't transconding |
| 20:34.39 | mjordan | I didn't say transcoding |
| 20:34.55 | mjordan | The mere act of looking up a media format type - in certain places *that is not just transcoding* - takes more effort |
| 20:35.17 | mjordan | If you don't like it, quite frankly, patches welcome. I'm not going to investigate something that is not a problem. |
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| 20:35.41 | ibercom | In weekend the CPU is 1.8 and the next weekend 5 o 7 % |
| 20:35.52 | ibercom | without calls. |
| 20:35.52 | mjordan | tell me when you hit 99%. |
| 20:36.28 | ibercom | Never, I get 50~60% |
| 20:36.52 | ibercom | With only 100 calls SIP and PRI |
| 20:38.33 | ibercom | In weekend the CPU is 1.8% and the next weekend 5 o 7 %, it is strange. |
| 20:39.36 | ibercom | I see problems with events.c and format_cap.c. I think so. |
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| 22:57.28 | Davlefou | Hi, is it possible to use agi for voicemail? |
| 22:59.06 | Davlefou | Yes, it is |
| 23:03.37 | newtonr | Davlefou, glad I could help |
| 23:06.14 | Davlefou | newtonr, i find the reply. |
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| 23:27.50 | Gazzonyx | So, now I'm just trying to be able to dial an extension and have that extension just record the input. I've created a [record] context in extensions as such: |
| 23:27.53 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 23:28.08 | Gazzonyx | [record] |
| 23:28.08 | Gazzonyx | exten => 1,1,Answer() |
| 23:28.08 | Gazzonyx | exten => 1,2,Record(recording%d:ulaw) |
| 23:28.08 | Gazzonyx | exten => 1,3,Hangup() |
| 23:29.14 | Gazzonyx | and added in sip.conf : [1](test) \n allowguest=yes \n context=record |
| 23:29.35 | Gazzonyx | That's all I should need to get this work work when I call extension 1, right? |
| 23:30.11 | [TK]D-Fender | Gazzonyx, no. |
| 23:30.32 | [TK]D-Fender | Gazzonyx, Ther is the4 [general] section which you are required to have for a sane sip.conf |
| 23:30.52 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-pfkywfykljvqbzfh) |
| 23:30.53 | [TK]D-Fender | Gazzonyx, And you haven't accounted for allowing codes properly, etc. No mention of passwords being set, etc |
| 23:31.07 | Gazzonyx | [TK]D-Fender: Ah, yes, I have that. I meant for this thing specifically. |
| 23:31.54 | *** join/#asterisk fang0654 (~danny@208.77.71.230) |
| 23:32.13 | fang0654 | Is there any way to automatically get packet captures of calls? So each call leaves a .pcap file? |
| 23:33.59 | [TK]D-Fender | Gazzonyx, You also have no TYPE or HOST specified for that peer |
| 23:34.17 | [TK]D-Fender | Gazzonyx, "allowguest" is NOT a vlid option for a device entry either |
| 23:34.21 | [TK]D-Fender | ~book |
| 23:34.22 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 23:35.33 | Gazzonyx | [TK]D-Fender: Okay. Thanks. I just started looking at the book under Configuring Asterisk. :) |
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| 23:41.43 | calum_ | Does anyone know of a trunk provider (sip or iax2) that provides encryption. particularly media encryption? |
| 23:51.12 | rrittgarn | @calum_ if you dont control both sides, can it ever be secure? As in what if your provider supports SRTP, but they are passing your call to the PSTN? Is the PSTN inherently secure? Not sure on the right answers, just something to think about |
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| 23:52.07 | WIMPy | I think the answer was ZRTP. |
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