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| 01:01.04 | vstemen | After upgrading to FreeBSD 11.1, asterisk was upgraded from 13.9.1 to 13.16.0. Now, using the same configuration that has been working all along to record calls into ogg files, it gives me this error |
| 01:01.06 | vstemen | WARNING[101508][C-00000003]: format_ogg_vorbis.c:385 int ogg_vorbis_seek(struct ast_filestream *, off_t, int): Seeking is not supported on OGG/Vorbis streams in writing mode! |
| 01:01.18 | vstemen | Anybody have any ideas? |
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| 01:19.54 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 04:34.44 | vstemen | In case anybody was seeing this. The ogg call recording problem is resolved. The error msg is still there but apparently it does not prevent it from recording. After installing the system, I forgot to put my /var/spool/asterisk/monitor sym linke back to point to where I keep my call recordings. So I was not looking in the right place. |
| 04:34.51 | vstemen | So it is recording fine. |
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| 07:28.44 | dawn | Hello, I am trying to setup my fxo using asterisk and dahdi. Whenever I try to place an external call I get the message that the dahdi channel cannot be created, and congested everyone is busy. I have been googling for a couple of days, if anyone could point me to a good place to start with configuring the fxo I would love them forever, thank you advance. |
| 07:34.01 | dawn | WARNING[21219][C-00000017]: app_dial.c:2431 dial_exec_full: Unable to create channel of type 'dahdi' (cause 0 - Unknown) |
| 07:34.09 | dawn | this is the warning I am recieving |
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| 08:25.02 | LiuYan | dawn: is the dahdi driver/module loaded in OS? issue`dahdi_tool` command in shell to see the status. |
| 08:33.07 | dawn | LiuYan I managed to blacklist the netjet module which was loaded first, and it did show up in dahdi_tool - however I am trying a clean install because I believe things are not sane in my configs |
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| 09:23.23 | cwhuang | Hi, I'm using Asterisk 15 with cdr_mysql backend |
| 09:23.55 | cwhuang | I found the start, answer, end are all set the same in a cdr record. Why? |
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| 09:24.10 | cwhuang | mysql> select src,dst,start,answer,end,duration from cdrtable; |
| 09:24.10 | cwhuang | +------+------+---------------------+---------------------+---------------------+----------+ |
| 09:24.14 | cwhuang | | src | dst | start | answer | end | duration | |
| 09:24.19 | cwhuang | +------+------+---------------------+---------------------+---------------------+----------+ |
| 09:24.22 | cwhuang | | 601 | 111 | 2018-02-04 17:02:10 | 2018-02-04 17:02:10 | 2018-02-04 17:02:10 | 14 | |
| 09:25.01 | cwhuang | I think the answer and end are incorrect. What happened? |
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| 09:48.00 | ChannelZ | dawn: what DAHDI hardware are you using? Are you sure the driver is running? Is it configured properly? |
| 09:49.17 | dawn | pci:0000:03:00.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F |
| 09:49.19 | dawn | pci:0000:05:01.0 wctdm- e159:0001 Wildcard TDM400P REV E/F |
| 09:49.30 | dawn | reported from dahdi_hardware ChannelZ |
| 09:50.36 | dawn | I have the netjet and dahdi_dummy in my blacklist |
| 09:51.12 | ChannelZ | And /etc/dahdi/system.conf ? And /etc/asterisk/dahdi-channels.conf ? |
| 09:52.33 | dawn | I have just reinstalled and have just this moment put those entries in modprobe.d/dahdi.conf beyond this I am still learning |
| 09:54.45 | ChannelZ | well so you must configure the hardware with /etc/dahdi/system.conf first so tell it what signalling to use for each channel. Then you configure those channels in asterisk with /etc/asterisk/dahdi-channels.conf |
| 09:54.50 | dawn | so now I am looking into /etc/dahdi/system.conf |
| 09:55.25 | dawn | noticed loadzone and defaultzone where wrong so corrected |
| 09:55.47 | dawn | it says fxsks=1 |
| 09:55.49 | ChannelZ | What modules are on your TDM400ps? |
| 09:56.22 | dawn | FXO |
| 09:57.12 | ChannelZ | Just a single module on each? |
| 09:57.26 | dawn | I only have the single module yes |
| 09:57.42 | dawn | second card has none ! |
| 09:58.37 | ChannelZ | ok well that one won't do you any good.. but anyway, so fxsks=1 is correct then (FXS signalling for an FXO port) |
| 09:59.06 | dawn | I shall remove the empty card |
| 10:00.46 | ChannelZ | So then in /etc/asterisk/dahdi-channels.conf you want something like (at minimum) |
| 10:00.53 | ChannelZ | signalling=fxs_ks |
| 10:01.12 | ChannelZ | context=pstn |
| 10:01.24 | ChannelZ | channel => 1 |
| 10:02.01 | dawn | ok confirm that this is so |
| 10:02.07 | ChannelZ | That will consume DAHDI channel 1 as FXS signalling inside asterisk |
| 10:03.03 | ChannelZ | Then on the asterisk console, "dahdi show channels" should show channel 1 if everything is right |
| 10:03.45 | dawn | pseudo default default Yes |
| 10:03.46 | dawn | <PROTECTED> |
| 10:03.55 | dawn | is the output |
| 10:04.27 | ChannelZ | ok so then what does your Dial() look like? |
| 10:05.03 | ChannelZ | You either have bad syntax or something is broken that it can't create the channel correctly |
| 10:06.01 | dawn | this is where I am up to configuring, I suspected that my configs was in a mess and that was the reason why I reinstalled so it would at least be sane |
| 10:07.09 | dawn | ChannelZ thank you :) |
| 10:07.41 | ChannelZ | ...? does that mean it works now or... |
| 10:08.15 | dawn | no it means I look into sorting my configs out |
| 10:08.49 | ChannelZ | Well I'm saying your dahdi configs _seem_ to be OK as the channel is showing up in asterisk.. |
| 10:09.00 | ChannelZ | My last question is, what does your test extension look like? |
| 10:09.11 | ChannelZ | That you're trying to get to dial out |
| 10:09.31 | dawn | oh sorry, I have tried several googled examples none to avail |
| 10:11.28 | ChannelZ | ok since you're not going to tell me... |
| 10:11.44 | dawn | I was just sftp from my backup to show you |
| 10:11.46 | ChannelZ | exten => 1234,1,Dial(DAHDI/1/1234567890) |
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| 10:12.13 | ChannelZ | where 1234 is some text extension and 1234567890 is some legitimate phone number |
| 10:12.41 | ChannelZ | assuming you have some other SIP phone/softphone or something setup that you're testing trying to make a call with |
| 10:13.49 | dawn | exten => _9.,1,Dial(dahdi/g0/${EXTEN:1}..To) this is from my previous setup |
| 10:14.51 | dawn | <PROTECTED> |
| 10:15.19 | dawn | if it is ok with you ChannelZ I would like to do so and then come and speak with you again shortly :) |
| 10:15.31 | ChannelZ | yea sure |
| 10:15.37 | dawn | thanks dude |
| 10:15.49 | ChannelZ | but how the hell were you testing it before? When you came in and pasted the 'cannot create channel' error? |
| 10:16.56 | dawn | I was on messy setup ubuntu 16 where my friend had removed pjsip because of problems and she replaced it with chan_sip |
| 10:17.24 | dawn | so I decided to reinstall and on ubuntu 17.04 and hoped it would be more sane |
| 10:17.44 | ChannelZ | oh.. so this was all after-the-fact |
| 10:18.33 | dawn | Kinda, that was probably my third install, all the previous installs was the same so was pulling my hair out |
| 10:19.16 | dawn | I figure it is how I am configuring sip and extensions |
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| 11:45.39 | dawn | ChannelZ Thank you, I managed to make an out going call but everything sounded like they/I were robots :) |
| 11:46.06 | ChannelZ | possibly/maybe the codec on your SIP channel |
| 11:46.21 | dawn | thank you will look into that now |
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| 12:26.20 | cwhuang | Checking the code, it explicitly treats start, answer, end the same. I can't believe it... |
| 12:26.42 | cwhuang | https://github.com/asterisk/asterisk/blob/15/addons/cdr_mysql.c line 279-285 |
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| 12:26.57 | cwhuang | Anyone can explain it? |
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| 12:31.16 | Samot | People still use cdr_mysql? |
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| 12:48.55 | Worldexe | Why not? |
| 12:50.43 | Samot | It has been deprecated since 1.i |
| 12:50.47 | Samot | Er 1.8 |
| 12:51.02 | Samot | So..it was replaced like 7 years ago. |
| 12:54.11 | Samot | The simple answer is "You should not use 1.8 configs in Asterisk 13 or 15" |
| 13:00.39 | Samot | Also, line 264: /* Construct SQL */ <-- That's important. |
| 13:01.08 | Samot | Because the lines following that "treat them the same" is the SQL formatting for insertion into the database. |
| 13:01.28 | Samot | So yeah, they are treated the same in that aspect because all database insertions should be. |
| 13:02.17 | Samot | cwhuang: Does that explain it? |
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| 13:19.28 | Worldexe | well, i use func_odbc and save call stats myself directly from dialplan |
| 13:19.45 | Worldexe | but what should i use instead of cdr_mysql? just in case |
| 13:22.53 | Samot | Res_odbc |
| 13:22.59 | Samot | The replacement |
| 13:23.36 | Worldexe | okay |
| 13:23.39 | Samot | Which is also the default method for Asterisk to talk to MySQL now |
| 13:23.45 | Samot | Since 1.8 |
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| 13:41.41 | dawn | Samot: I thought Asterisk used sqlite3 |
| 13:42.08 | sibiria | it does for its internal DB |
| 13:43.00 | sibiria | and if you wish, you can make use of all-asterisk functionality to use that key-value store for CDRs |
| 13:58.44 | dawn | Is there any known issues with the ulaw codec? My PSTN line keeps breaking up. I'm planning on replacing the rj11 cable to see if it relieves the issue. |
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| 14:26.39 | sibiria | most likely just network latency |
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| 14:38.35 | dawn | sibiria: The LAN is 1gbit/s |
| 14:40.29 | Samot | The lan is not what the PSTN uses |
| 14:43.28 | dawn | Agreed it's not |
| 14:43.44 | dawn | and I thought sibiria was refering to my LAN |
| 14:49.59 | cwhuang | Samot: thanks. then what is the config for Asterisk 13 or 15? |
| 14:53.53 | cwhuang | Samot: what config did you mean exactly? cdr_mysql.conf or anything else? |
| 14:56.03 | Samot | Cdr_mysql is deprecated |
| 14:56.25 | Samot | It was replaced by res_odbc like 7 years ago |
| 14:56.54 | Samot | Which is the defaul way * talks to mysql |
| 14:58.10 | Samot | Plus the code you pointed to was about inserting the cdr data to the database...not the data itself |
| 15:00.14 | cwhuang | Samot: I'm not familiar with the code enough. I think it sets all the value of start, answer, end to cdr->start. Isn't it? |
| 15:00.46 | Samot | No |
| 15:00.58 | Samot | Thats for putting it _in_ |
| 15:01.06 | dawn | Muse |
| 15:01.21 | Samot | Once all the data has been collected |
| 15:04.29 | cwhuang | Samot: then what does line 290 do? tm is set to cdr->start, no matter the field is start, answer, end. It doesn't look right. |
| 15:05.00 | Samot | It is for database insertion |
| 15:05.17 | Samot | All the data gets inserted the same way |
| 15:05.27 | cwhuang | Samot: if not, please explain why they are the same in my cdrtable? |
| 15:05.40 | Samot | I dont know |
| 15:06.02 | Samot | Mysql_cdr is no longer used as the default method |
| 15:06.10 | Samot | I havent used it in years |
| 15:06.26 | Samot | Because its not recommended |
| 15:08.14 | cwhuang | That doesn't explain the code, except it's unmaintained |
| 15:09.52 | cwhuang | Samot: are you the developer? |
| 15:10.28 | file | it looks like a bug that was introduced in the past few months |
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| 15:11.11 | cwhuang | file: are you talking to me? thanks! I think so. |
| 15:11.27 | file | but I doubt anyone would touch it |
| 15:14.12 | cwhuang | file: ok, I see. thank you! |
| 15:14.33 | Samot | No am i not. |
| 15:14.40 | Samot | And a bug was my next option |
| 15:14.52 | Samot | And as i pointed out, since its deprecated... |
| 15:15.07 | Samot | As file said, might not get touched |
| 15:15.47 | Samot | The answer is to try res_odbc |
| 15:15.53 | Samot | If you are on 15 |
| 15:17.18 | cwhuang | Samot: ok, thank you anyway |
| 15:17.32 | file | we don't even build it by default |
| 15:17.34 | cwhuang | Indeed it's my first time to try 15 |
| 15:19.17 | cwhuang | file: I just installed Fedora's package, asterisk-mysql |
| 15:19.37 | Samot | Yeah... |
| 15:19.44 | Samot | Thats probably why |
| 15:19.44 | file | the project doesn't control the upstream packages so it is certainly possible they made a package for it |
| 15:19.58 | Samot | You have had to make extra effoet to install it for years |
| 15:20.00 | file | project wise it's not built by default, it's marked as deprecated, and menuselect tells you to use cdr_adaptive_odbc |
| 15:22.04 | Samot | I have noticed that a lot of upstreams have seriously lagged on removing and replacing deprecated items. |
| 15:22.17 | Samot | Honestly it seems some just stopped at 11 |
| 15:23.57 | Samot | I think PJSIP scares them :) |
| 15:24.26 | Samot | FOP2 is a great example. Great software.. |
| 15:24.40 | Samot | Does all the things you really need from a call management point... |
| 15:24.56 | Samot | But it's two biggest flaws right now. hasn't moved from Monitor to MixMonitor |
| 15:25.05 | Samot | And ZERO PJSIP support. |
| 15:25.27 | file | I've found that a lot of projects built on Asterisk just don't follow what Asterisk is doing or try to participate |
| 15:25.35 | Samot | That too. |
| 15:25.44 | Samot | But FOP2 is a bit different. |
| 15:26.04 | Samot | They just haven't updated their software with new changes. |
| 15:26.31 | Samot | But yeah, anyone using Asterisk as their underlying voice engine and still haven't gotten to 13... |
| 15:26.40 | Samot | Are just not caring. |
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| 16:39.16 | fbnts | Hi, I use a page function to multiple Grandstream SIP phones to play a file and hangup. It works great but the handset initiating the page passes it's audio through to the handsets. (This is used in a commercial kitchen to signal to staff that food is ready to go out). Is there a way to mute the call audio either in the Dial() or Page() commands? |
| 16:40.32 | Samot | Page has a duplex setting |
| 16:40.37 | Samot | I believe |
| 16:40.44 | Samot | That needs to be off |
| 16:41.02 | Samot | Or see if the phone has a setting as well |
| 16:41.18 | Samot | But youre opening a two channel basically |
| 16:41.26 | Samot | Way+ |
| 16:42.18 | fbnts | Samot: yes the 'd' option to set duplex. I don't call that option so it's one-way audio but from kitchen to all phones |
| 16:43.19 | fbnts | oh, I've just found the MUTEAUDIO() function - I'll give that a try |
| 16:44.57 | sibiria | i have a question about a weird tendency i see with multiple operators we rent trunks from for outgoing calls |
| 16:45.17 | sibiria | we often hear our outgoing audio stream in the incoming stream coming back from the recipient of our calls |
| 16:45.26 | sibiria | in the background |
| 16:46.04 | sibiria | are they using some garbage joint stereo encoding which is then passed on back to us? |
| 16:48.22 | sibiria | like, the absolutely worst of the worst among mp3 encoders in joint stereo mode or so |
| 16:50.10 | sibiria | anyone ever noticed anything similar? |
| 16:54.48 | Samot | So you make an outbound call |
| 16:55.39 | Samot | And you are hearing echo back of your voice inside the reciever speaker? |
| 16:56.18 | sibiria | when the incoming audio from the recipient of the call goes up a bit, we hear our outgoing audio stream in the background, yes |
| 16:56.40 | Samot | All devices? |
| 16:56.42 | sibiria | (no speaker, this is strictly computer side) |
| 16:56.49 | sibiria | no devices |
| 16:57.01 | Samot | So these are softphones? |
| 16:57.29 | sibiria | no devices of any kind; app to app |
| 16:58.15 | sibiria | there's no analog component involved |
| 16:58.19 | Samot | Echo suppression, cancellation are things the endpoints do |
| 16:58.55 | sibiria | i mean, we hear this when the recipient is an operator voicemail |
| 16:59.05 | sibiria | it's not an analog feedback problem |
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| 16:59.16 | sibiria | no speaker->microphone blooper |
| 17:00.00 | Samot | I didnt say it was analog problem |
| 17:00.22 | Samot | Echo suppression, cancellation are voip settings |
| 17:00.35 | sibiria | ok so just a bit of dodgy software on their end |
| 17:00.38 | Samot | Sip devices/softphones have them |
| 17:00.52 | Samot | If neither app is doing it..m |
| 17:01.15 | sibiria | there's no way to indicate the desire to have this turned off via SIP or SDP i guess? |
| 17:01.15 | Samot | You are hearing echo |
| 17:01.19 | Samot | Thats you |
| 17:01.43 | Samot | Its part of the stream |
| 17:01.57 | Samot | Those methods clean it up |
| 17:02.48 | sibiria | right, i understand that it's a best effort deal trying to counter for eventual feedback |
| 17:03.40 | sibiria | we have none of this configured or set up on our end, so i assume there's no way for us to ask the opposite end per session to turn it off |
| 17:04.08 | Samot | Its on your side |
| 17:04.13 | Samot | They cant do anything |
| 17:04.16 | Samot | You have to |
| 17:04.29 | Samot | If this is an app, you need to address it |
| 17:04.36 | sibiria | it's just asterisk |
| 17:04.37 | sibiria | nothing else |
| 17:05.10 | sibiria | and we've done no config of our own to enable this - if asterisk does it at all, or by default, i don't know |
| 17:06.33 | sibiria | we place outgoing calls via asterisk directly. we record the two streams using MixMonitor. nothing else involved |
| 17:07.23 | Samot | So youre not using an app really |
| 17:07.40 | Samot | Youre using Originate |
| 17:08.14 | sibiria | i'm merely pointing out that we don't use anything phone, soft or hard, and that this manifests even when no recipient picks up the call :) |
| 17:08.30 | sibiria | (as in, their voicemail answers) |
| 17:09.00 | sibiria | and that it's not isolated to one trunk provider, but shows up in, well, all of them |
| 17:09.16 | sibiria | does asterisk do any echo cancellation? |
| 17:09.16 | Samot | Well |
| 17:09.20 | Samot | No |
| 17:09.50 | sibiria | hence my theory of some rubbish joint stereo encoding of the two legs by the provider, and feeding us that |
| 17:10.03 | Samot | No |
| 17:10.09 | Samot | Thats not how it works |
| 17:10.22 | Samot | There is no stereo on the PSTN |
| 17:10.28 | sibiria | i know |
| 17:10.31 | sibiria | two legs - us, them |
| 17:10.43 | Samot | So providers dont do that |
| 17:10.51 | syadnom | excuse the repeat, hoping someone new is watching... |
| 17:10.54 | syadnom | wondering about asterisk and real-time mysql latency. |
| 17:10.57 | Samot | Your leg is between the provider and yoy |
| 17:10.58 | syadnom | specifically, using mysql router to split reads/write between a local slave and a remote master. |
| 17:10.59 | Samot | You |
| 17:11.02 | Samot | Thats it |
| 17:11.15 | syadnom | assuming the connections stay up and there is no packet loss, can asterisk play well with a database write <=200ms away? |
| 17:11.19 | Samot | Everything after that is the provider to whereever they send the call |
| 17:12.18 | sibiria | i suppose i can assume the receiving operator is doing it per default |
| 17:12.29 | sibiria | as in recipent's opeartor |
| 17:13.19 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 17:15.34 | sibiria | it's not really a problem, it's just something that catches my attention whenever i'm reviewing audio quality in the recordings |
| 17:20.21 | sibiria | syadnom: i'd find it hard to believe there'd a timeout either on connecting to the dbms or on finalizing the query that would be set as low as 200ms in this context |
| 17:20.58 | syadnom | sibiria, what do you mean, set as low as 200ms? |
| 17:21.15 | sibiria | syadnom: you're asking if there is a problem with 200ms latency between asterisk and your DBMS |
| 17:21.43 | sibiria | verifying RSA f.e. over a TLSd connection can take more than 0.2 seconds (subject of connection timeout) |
| 17:21.49 | sibiria | a query alone can take more than 0.2 seconds |
| 17:22.10 | sibiria | enforcing a timeout that low, on data that isn't completely discardable, would be very unwise |
| 17:22.23 | syadnom | I'm not enfocing a timeout |
| 17:22.46 | sibiria | i know - but you're asking if something with asterisk is, indirectly :) |
| 17:23.04 | syadnom | I'm wondering if adding 200ms of transport latency is an issue. and no tls, established vpn tunnel. |
| 17:23.10 | sibiria | no it shouldn't be |
| 17:24.35 | *** join/#asterisk jkroon__ (~jkroon@165.16.204.170) |
| 17:25.21 | sibiria | unless you plan on a very high frequency of queries mid-call or so? |
| 17:25.31 | sibiria | f.e. verbose log output to dbms? |
| 17:25.35 | syadnom | you just cut my though lol. |
| 17:25.42 | sibiria | yeah don't do that |
| 17:25.43 | syadnom | yes, lots of queries |
| 17:25.49 | sibiria | it's a very unwise setup imo |
| 17:26.09 | syadnom | the pbx software I'm using has some heavy queries |
| 17:26.53 | syadnom | not a big deal with a local mysql server on an SSD, but other users are claiming instability and having to restart asterisk a few times *per day* with ~50ms latency to the db server. |
| 17:28.04 | sibiria | if these queries happen very frequently during the flow of the dial plan i can see how it would stall the call |
| 17:28.13 | syadnom | there are some writes, but they are lightweight (in comparison), it's the heavy read queries. So I'm trying to vet out the idea of using mysql router to split reads to a local slave and writes to the remote master. This to overcome the limit of master-master to two sites (and refusing to do circular replication) |
| 17:28.17 | sibiria | but restarting asterisk, that sounds really of |
| 17:28.34 | sibiria | you keep your dial plan in the DB? |
| 17:28.38 | syadnom | yes |
| 17:28.46 | sibiria | yeah we had big problems with that, too |
| 17:28.59 | sibiria | we have had to render it to file from DB regularly instead |
| 17:29.03 | sibiria | and reload the dial plan |
| 17:29.13 | sibiria | moving prio to prio, or context to context, began slowing calls down tremendously |
| 17:29.16 | syadnom | peers can register to multiple asterisk servers and make calls between servers. |
| 17:29.20 | sibiria | even with indices |
| 17:30.02 | sibiria | we have a few thousand contexts active at any moment, each with 10+ priorities |
| 17:30.15 | sibiria | even with indexing it just didn't work out for asterisk |
| 17:30.25 | sibiria | i do not recommend this setup |
| 17:30.46 | sibiria | and, really, we had this problem even with the db being a few ms away on the network |
| 17:31.29 | syadnom | don't have an issue locally with thousands of peers. |
| 17:31.40 | sibiria | doing stuff like ocassional db access now and then during the flow, f.e. during dialing or hangup etc., isn't a problem at all |
| 17:37.25 | syadnom | what other users are telling me is that the number of queries made by asterisk including some heavy AEL compounds from db latency. So 50ms really means 500ms of threads stalled which asterisk presents as high load. |
| 17:37.56 | sibiria | yes i can imagine. high latency and high frequency is a problem |
| 17:38.18 | sibiria | you'll be delaying the flow of the call incrementally |
| 17:38.41 | sibiria | is there a problem with rendering to file regularly from the main db? |
| 17:38.49 | sibiria | or using streaming replication to a local dbms? |
| 17:39.06 | syadnom | yes, having peers hit multiple asterisk servers means a contantly updating dialplan. |
| 17:39.47 | sibiria | your setup really requires manipulating the dial plan in real time? |
| 17:40.19 | syadnom | if you dial a peer that is on another asterisk server, the database has to be aware if it's registration there. |
| 17:40.34 | syadnom | so yeah, part if this is finding the peer. |
| 17:41.00 | syadnom | and the problem with replication is the requirement to write to sipfriends and other tables. which means master-master replication. |
| 17:41.35 | syadnom | which implies a limit of 2 sites/2 mysql masters. Wouldn't touch circular replication for anything. adds a big reliability hole. |
| 17:42.20 | sibiria | that's what things like Galera is for |
| 17:42.31 | sibiria | but that won't really save you if you have 200ms latency between the nodes |
| 17:42.35 | syadnom | galera dies horribly with high latency |
| 17:43.30 | syadnom | master-master works like a champ. the likelyhood of a peer registering remotely and a person dialing that peer within a second or two is incredibly low. |
| 17:44.31 | syadnom | I'm exploring multi-source replication but I'm not sure how that will work out. I've never used it, it adds complexity, and I don't know how this pbx system will handle gtid or if I'll get support with an 'exotic' database setup. |
| 17:46.23 | *** join/#asterisk paulgrmn_ (~paulgrmn@184.75.212.68) |
| 17:47.10 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 17:59.02 | sibiria | it adds risk more than complexity. multi-master replication across several nodes can operate safely but it requires quite some care even when using offset auto-incrementing keys etc. |
| 17:59.14 | sibiria | i hope you manage to solve the tricky situation |
| 18:02.55 | Worldexe | db latency is a problem; i lowerer overall asterisk system load 2x-3x times by just switching to writing into intermediate dedicated fast db instead of quite a loaded production |
| 18:03.14 | syadnom | @Worldexe, how did you do that? |
| 18:13.45 | sibiria | local node i suppose? |
| 18:14.11 | sibiria | it's also worth pointing out that you can control galera's synchronous behavior |
| 18:14.36 | sibiria | to sacrifice part of the data safety for faster performance |
| 18:14.50 | syadnom | galera always always dies with high latency. I doubt you'll find someone who tried it and it didn't die a horrible death. |
| 18:15.25 | sibiria | i can't say i've found this to be a problem - and i've run galera clusters across WAN - across nation borders - with more than a little latency |
| 18:15.32 | sibiria | with high write load, without accident |
| 18:21.10 | syadnom | for asterisk real-time? |
| 18:22.24 | sibiria | no, but that's besides the point. *galera* manages clusters with high latency just fine |
| 18:22.33 | sibiria | if this or that application doesn't is that application's fault |
| 18:23.14 | sibiria | you said that galera crashes when there's high latency between nodes or between application and node, and i can't help feeling that's not right |
| 18:23.16 | syadnom | which version of galera I guess, I've looked at the percona version |
| 18:23.30 | sibiria | i've only worked with galera for mariadb |
| 18:24.03 | sibiria | (which in turn does use percona xtradb, but it's still mariadb) |
| 18:24.33 | syadnom | ok, well I'm going to do some research on that make sure my notions are valid or not |
| 18:59.45 | *** join/#asterisk gzp (~gzp@catv-80-99-9-120.catv.broadband.hu) |
| 19:01.11 | gzp | Executing [failed@messages:1] NoOp("OutgoingSpoolFailed", "Reason: 0") in new stack |
| 19:01.27 | gzp | what is OutgoingSpoolFailed? I can't find in my configuration |
| 19:09.02 | Samot | It's your dialplan. |
| 19:09.53 | Samot | Or is this some system that generates the dialplan for you? |
| 19:10.58 | gzp | yes, MessageSend generate it |
| 19:11.16 | gzp | but always exits with non-zero status, so it fails |
| 19:11.23 | gzp | I'm unable to determine whats the problem |
| 19:11.32 | gzp | messaging works anyway |
| 19:13.29 | Samot | That NoOp() line had to be written by someone. |
| 19:14.29 | sibiria | gzp: the bottom line is that your outgoing call failed |
| 19:16.33 | Samot | I'm not sure. |
| 19:16.36 | Samot | It's not a call. |
| 19:16.37 | sibiria | you could expand on that NoOp (which should be a Verboe ideally, it's not recommended to use NoOp for debug output) by having it show the disposition etc. |
| 19:16.53 | Samot | And 0 is not a valid response to MessageSend(). |
| 19:17.05 | Samot | That's the thing... |
| 19:17.15 | Samot | 2:01:09 PM G<gzp> Executing [failed@messages:1] NoOp("OutgoingSpoolFailed", "Reason: 0") in new stack |
| 19:17.15 | Samot | 2:01:25 PM G<gzp> what is OutgoingSpoolFailed? I can't find in my configuration |
| 19:17.27 | Samot | ^^ Doesn't know what that is but sees it in the log. |
| 19:17.45 | Samot | So either someone wrote that in the company and it's not realized |
| 19:18.02 | Samot | Or they have a system that was written by a third party and that exists in the dialplan for a reason. |
| 19:18.23 | Samot | But again, 0, is not a valid returned status for MessageSend() |
| 19:19.15 | gzp | hmm |
| 19:19.27 | Samot | gzp: Are you running a pure Asterisk install? |
| 19:19.40 | gzp | yes, with almost empty config |
| 19:19.41 | Samot | gzp: Or is this some appliance or Distro that _uses_ Asterisk? |
| 19:19.44 | Samot | OK. |
| 19:19.44 | gzp | no |
| 19:19.45 | Samot | So |
| 19:19.48 | gzp | compiled myself |
| 19:19.58 | Samot | That messages context you wrote? |
| 19:20.11 | gzp | yes |
| 19:20.15 | Samot | That NoOp() message you pasted.. |
| 19:20.21 | Samot | Had to be written by you. |
| 19:20.37 | gzp | no, thats not! |
| 19:20.43 | gzp | thats the problem |
| 19:20.46 | Samot | failed <-- that would be an extension you wrote in [messages] |
| 19:20.54 | Samot | So you don't have a context called [messages]? |
| 19:21.01 | znoteer_ | I had a working voip system Friday. Made a couple of calls. Received a couple of calls. Then, I get another call, and there is no sound either way. Since, that's the case. I can call out and people can call in, but when the line gets answered, no one can hear anything? There were no upgrades, tunings or other changes to anything on my end. I'm not seeing any errors on the console. Where do I |
| 19:21.03 | gzp | never defined OutGoingSpool |
| 19:21.07 | znoteer_ | start to track this down? |
| 19:21.12 | Samot | So you don't have a context called [messages]? |
| 19:21.16 | gzp | Samot, I have that context |
| 19:21.18 | Samot | OK |
| 19:21.24 | Samot | NoOp()'s are for you |
| 19:21.28 | Samot | They are "comments" |
| 19:21.31 | gzp | ah |
| 19:21.37 | Samot | so if you have a NoOp() message |
| 19:22.06 | Samot | That means someone wrote NoOp(OutgoingSpoolFailed", "Reason: 0") in the dilplan |
| 19:22.09 | Samot | dialplan |
| 19:22.22 | Samot | Or that 0 is a ${VAR} |
| 19:22.31 | Samot | But someone had to write that line. |
| 19:22.41 | sibiria | but, uh... it's pbx_spool that sets the channel to OutgoingSpoolFailed when it, well, fails |
| 19:22.42 | gzp | let me pastebin, hold on |
| 19:22.49 | Samot | OK. |
| 19:22.54 | Samot | Show the debug. |
| 19:22.56 | Samot | That would help. |
| 19:23.32 | gzp | https://pastebin.com/SHXi76TW |
| 19:25.28 | gzp | https://pastebin.com/LpReRV80 |
| 19:26.17 | gzp | another problem I can't reproduce the problem right now. Disconnected the another part where to I try send the text message, but asterix still store it somewhere in the memory and says SUCCESS for the delivery |
| 19:26.30 | gzp | can I disable this feature? |
| 19:27.43 | Samot | How are you sending this? |
| 19:28.18 | gzp | via sip? |
| 19:28.28 | Samot | Let me try that again. |
| 19:28.35 | Samot | How does Asterisk receive the message? |
| 19:28.57 | gzp | from zoiper via sip |
| 19:29.08 | Samot | OK, SIP SIMPLE. |
| 19:30.03 | Samot | Show this from the start. |
| 19:30.11 | Samot | Where it first comes in. |
| 19:30.47 | gzp | https://pastebin.com/NDVB6B0d |
| 19:31.49 | Samot | Where is the failedmsg portion of this dialplan? |
| 19:31.59 | Samot | Because that looks like it was sent. |
| 19:33.02 | *** join/#asterisk visip (~visix@gateway/tor-sasl/visip) |
| 19:33.31 | gzp | https://pastebin.com/ZhYDrykM |
| 19:38.03 | Samot | I really hope that from line 39 and on in this that was a pasting error are your part. |
| 19:38.29 | Samot | exten => _X.,n,Hangup()[app-fakeanswer] |
| 19:38.54 | gzp | let me check |
| 19:39.05 | *** join/#asterisk areski (~areski@37.223.2.207) |
| 19:39.24 | gzp | no :-o |
| 19:39.35 | Samot | So it was a pasting error... |
| 19:39.42 | Samot | Or is it in the dialplan like that? |
| 19:39.48 | gzp | it is the dialplan |
| 19:39.57 | Samot | Then it's wrong. |
| 19:40.22 | Samot | You copy and pasted your fake answer app in the middle of your messages context. |
| 19:40.53 | gzp | ahh |
| 19:40.53 | Samot | 2:38:28 PM S<Samot> exten => _X.,n,Hangup()[app-fakeanswer] <-- That's wrong |
| 19:40.56 | gzp | I see |
| 19:40.59 | Samot | Yeah. |
| 19:41.00 | Samot | So.. |
| 19:41.06 | Samot | Not to completely nitpick here |
| 19:41.11 | Samot | 1) You did not write this. |
| 19:41.26 | Samot | 2) You need to tell us upfront when you've used random code from the Internet. |
| 19:41.56 | gzp | true |
| 19:43.38 | gzp | and why this dialplan success even if remote peer isn't alive? |
| 19:44.29 | Samot | That code was designed for it. |
| 19:44.37 | Samot | Do you even have the .sh file? |
| 19:44.44 | gzp | yes |
| 19:44.47 | Samot | OK. |
| 19:45.34 | gzp | and where is the message right now? in the memory? |
| 19:45.49 | gzp | if remote goes online, receives the message |
| 19:45.50 | Samot | Where do you store the outgoing files? |
| 19:46.10 | gzp | this isn't queued into file |
| 19:46.22 | Samot | That code, if the device is not available, generates a Callfile that sits in the Outgoing Spool... |
| 19:46.37 | Samot | It attempts to deliver that message a few times... |
| 19:46.52 | gzp | yes, that was working a few hours ago :-) |
| 19:46.57 | Samot | If it can't, as in the device doesn't come back up, it deletes it. |
| 19:47.13 | gzp | but now dialplan says SUCCESS even if the remote isn't online/registered |
| 19:47.14 | Samot | You have bad dialplan code |
| 19:47.33 | gzp | ah |
| 19:48.04 | gzp | thought I should redirect via MESSAGE_SENT_STATUS |
| 19:48.13 | gzp | s/via/based |
| 19:48.43 | Samot | SUCCESS - Successfully passed on to the protocol handler, but delivery has not necessarily been guaranteed. |
| 19:48.58 | Samot | In other words |
| 19:49.07 | Samot | It sent the message to sip:6001 |
| 19:49.11 | Samot | To be processed |
| 19:49.17 | Samot | The failure is happening after that |
| 19:49.27 | gzp | ok |
| 19:49.31 | Samot | Which means that it's not online and it failed to find a location for that pper |
| 19:49.34 | Samot | Which means that it's not online and it failed to find a location for that peer. |
| 19:49.39 | Samot | You probably need to clean this up. |
| 19:49.44 | Samot | It has errors |
| 19:49.49 | Samot | And well, is old. |
| 19:50.03 | gzp | the dialplan? |
| 19:50.12 | Samot | You have dialplan errors. |
| 19:50.17 | Samot | From how you added stuff. |
| 19:50.20 | Samot | That needs to be fixed |
| 19:50.23 | Samot | The code itself.. |
| 19:50.24 | gzp | started asterisk today |
| 19:50.35 | Samot | This example you got from the Internet.. |
| 19:50.40 | Samot | Is from years ago |
| 19:50.50 | Samot | Asterisk 10 |
| 19:50.59 | gzp | I understand it |
| 19:51.12 | Samot | So it could probably do with some tweaking. |
| 19:51.32 | gzp | is there a newer example somewhere? |
| 19:51.56 | gzp | or what should I fix? |
| 20:01.35 | Samot | I don't think there is. |
| 20:01.43 | Samot | I don't know what you should fix. |
| 20:02.04 | Samot | That example is wrapped around what the person who originally wrote was doing. |
| 20:02.14 | Samot | And people latched on to it |
| 20:02.26 | Samot | Either used it as is, as there are numerous sites that have it as an example. |
| 20:02.31 | gzp | is there any way to check the remote peer is online or offline? |
| 20:02.34 | Samot | Or made tweaks. |
| 20:02.41 | Samot | sip show peers |
| 20:02.45 | Samot | In the Asterisk cli. |
| 20:04.39 | gzp | ahh, monitored/unmonitored |
| 20:04.58 | gzp | AST thinks both clients are online (unmonitored) |
| 20:05.29 | gzp | so I think I should monitor my clients on lan? |
| 20:24.09 | znoteer_ | I'm trying to figure out why I suddenly have no sound going in or out on an iax2 connection. Tcpdump appears to be showing traffic going through the firewall to my asterisk box. core set debug 10, core set verbose 10, iax2 debug on doesn't show anything suspicious on the console. I've tried restarting my switch and my ISP's adsl converter box, but that hasn't helped. What else might I look at? |
| 20:33.38 | Samot | rtp set debug on <-- See if RTP is actually flowing during the call |
| 20:34.46 | znoteer_ | is rtp used for iax2? I thought everything passed through port 4569 for iax |
| 20:36.14 | Samot | I don't know, honestly. |
| 20:36.18 | Samot | I don't IAX2. |
| 20:38.31 | sibiria | it uses only one port, only udp traffic, but it might speak RTP somewhere in there |
| 20:38.42 | znoteer_ | ya, it seems that not many people do. I've been having a bit of an adventure with voip. Things worked peachy over the PSTN. |
| 20:39.28 | znoteer_ | ok, so I'll try and see of rtp debugging reveals anything |
| 20:40.12 | Samot | IAX2 is not a standard. |
| 20:40.13 | Samot | At all. |
| 20:40.28 | Samot | There is Asterisk and a small amount of systems that support it. |
| 20:40.31 | Samot | A very small amount. |
| 20:41.00 | sibiria | it's spencer's brainchild, made for asterisk, to make things easier |
| 20:41.15 | znoteer_ | should I ramp up core debug and core verbose while rtp debugging is on, or no need? |
| 20:41.22 | Samot | In a time when it was needed. |
| 20:41.28 | Samot | It's not really needed anymore. |
| 20:45.15 | znoteer_ | rtp set debug on did not reveal anything new |
| 20:47.21 | znoteer_ | more accurately, it didn't show anything happening rtp wise. But, I'm pretty sure rtp doesn't come into play with iax |
| 20:51.52 | Samot | Yes. |
| 20:51.55 | Samot | Yes it does. |
| 20:52.02 | Samot | IAX supports the same codecs as SIP |
| 20:52.07 | Samot | Well for the most part. |
| 20:52.17 | Samot | so IAX is still doing ulaw. |
| 20:52.26 | Samot | That means IAX is sending some sort of SDP |
| 20:52.41 | Samot | There has to be some method of the media going out. |
| 20:52.46 | Samot | RTP = Real Time Protocol. |
| 21:03.48 | znoteer_ | Everything I've read says that you only need to open a single port for iax: data and signalling use the same connection. RTP uses its own port range doesn't it? Could this SDP be sent via means other than RTP? |
| 21:08.06 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 21:08.24 | znoteer_ | oh wait, if this has to do with codec negociation, I can see that happening just fine when I have debugging turned on, so I don't think that could be the issue, though I am in over my depth |
| 21:15.06 | ChannelZ | iax2 is single port yes |
| 21:16.42 | znoteer_ | ChannelZ: do you know if that means no RTP? |
| 21:16.48 | ChannelZ | "iax2 set debug on" on the console and then making a call may or may not give you any insight as to what is happening in your case codec-wise |
| 21:17.38 | ChannelZ | It's not rtp separately like for SIP, no.. |
| 21:18.09 | *** part/#asterisk Sprinterfreak (~irc@p50857806.dip0.t-ipconnect.de) |
| 21:18.17 | znoteer_ | I've already tried iax2 debug. It appears to be selecting ulaw with no issues |
| 21:18.33 | Samot | RTP still has to flow |
| 21:18.42 | Samot | There has to be media. |
| 21:18.56 | Samot | Does the provider see media from you when the call connects? |
| 21:19.04 | ChannelZ | Yes it's just not over a separate port as RTP only data like SIP is what I'm saying. IAX2 muxes everything into a single UDP stream |
| 21:19.08 | znoteer_ | the media travels over the same port |
| 21:19.36 | ChannelZ | If IAX is connected at all, the audio is going over the same path and should be going as well. |
| 21:20.07 | ChannelZ | What is the actual path of the call? IE: SIP Phone -> Asterisk -> IAX2 to someplace else -> ??? |
| 21:20.16 | znoteer_ | https://paste.debian.net/hidden/bae9df0a/ |
| 21:20.37 | znoteer_ | thats debug 10, verbose 10 and iax2 debug on |
| 21:20.56 | znoteer_ | There's an incoming then an outgoing call in that paste |
| 21:21.50 | znoteer_ | incoming calls from my provider to asterisk to a pstn card to a pstn phone set |
| 21:22.22 | znoteer_ | outgoing calls from the same phone set to the telephony card, to asterisk to my itsp |
| 21:26.56 | ChannelZ | I see your incoming answers and does a Read(), you dont' hear that prompt when you test? (and what are you calling in with? Mobile phone or something else known working) |
| 21:27.43 | znoteer_ | when I call in with my cell phone, I hear the prompt, select an option, then hear nothing more |
| 21:28.21 | ChannelZ | so then it sounds more like your analog card is the problem, not IAX |
| 21:29.33 | ChannelZ | IE asterisk is getting audio out (and presumably in.. you could do an Echo() test and make sure) between it and your ITSP |
| 21:30.06 | ChannelZ | But for some reason you're not getting audio on your phones over the DAHDI channels |
| 21:30.23 | znoteer_ | the audio doesn't go either way |
| 21:30.55 | znoteer_ | could this be the card getting into some state that a reboot might fix? |
| 21:31.05 | ChannelZ | right.. but that's between your phones and the card/asterisk |
| 21:31.21 | znoteer_ | oh, right |
| 21:32.08 | znoteer_ | what about a reboot? Overkill? Time wasted? or possible solution? |
| 21:32.33 | ChannelZ | Have you tried setting up a dest extension that just dials one of your DAHDI channels to the other? Does that audio work? |
| 21:32.41 | ChannelZ | *test extension, not dest |
| 21:33.17 | znoteer_ | I happen to have to modules for phones on that card. Let me test audio between those two internal sets |
| 21:33.24 | ChannelZ | If the hardware is in a weird state and is indeed the problem, sure a reboot might sort that out. |
| 21:33.25 | znoteer_ | two modules |
| 21:34.19 | ChannelZ | suffice to say your IAX link its self doesn't seem to be the problem |
| 21:35.49 | znoteer_ | I just tried to call one internal phone to the other. No sound. Tried calling back the other way, no sound. So, I think I may try to reboot the box and see if that helps |
| 21:35.56 | ChannelZ | yeah |
| 21:37.00 | ChannelZ | bbl, lunch |
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| 21:55.29 | znoteer_ | ChannelZ: I hope lunch was tasty. Thanks for the help troubleshooting. I learned something which is always nice. I found the problem problem. It was one of my phones. Shortly after trying the internal phone to phone test, the one that I've been doing most of the testing with started to make funny noises. I tested between my cell and the other extension, and voice worked. So, I've replaced the |
| 21:55.35 | znoteer_ | malfunctionning phone with another I had lying around, and I'm in business |
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| 22:34.40 | ChannelZ | Good deal znoteer, glad you got it working |
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