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| 01:22.50 | DannyA | is there a variable that can be used in a dialplan which is the last incoming caller ID to a channel/extension? |
| 01:23.04 | DannyA | im trying to create a feature code similar to *69 |
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| 01:32.04 | Samot | No. You would have to store that information somewhere and then call on it. |
| 01:38.19 | DannyA | @Samot gotcha. so the flow would basically be, for this particular feature code, lookup in the CDR records the last incoming call to that extension and then call it? |
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| 01:40.51 | Samot | Just store it in AstDB |
| 01:41.04 | Samot | No need to make it complicated |
| 01:43.27 | DannyA | ok cool, will lookup how to do that. thanks! |
| 01:44.09 | Samot | You should be looking at FreePBX |
| 01:44.30 | DannyA | just found this! https://eric.lubow.org/2007/asterisk-69-with-14x/ |
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| 04:26.09 | karelk | I want to log cdr to MariaDB |
| 04:26.21 | karelk | what do I need to put into odbcinst.ini ? |
| 04:26.56 | karelk | on old Asterisk, where I logged to Mysql, I had: |
| 04:26.59 | karelk | [MySQL] |
| 04:26.59 | karelk | Description = MySQL ODBC MyODBC Driver |
| 04:26.59 | karelk | Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so |
| 04:27.27 | karelk | I assume this has to be different for MariaDB |
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| 05:27.01 | karelk | I have a problem: |
| 05:27.19 | karelk | when I start Asterisk, the log shows: NOTICE[2478] loader.c: 331 modules will be loaded. |
| 05:27.42 | karelk | but when I do 'module show', there are no modules |
| 05:27.49 | karelk | and asterisk does not work |
| 05:28.13 | karelk | I don't see any errors in the log that would explain why modules are not loaded |
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| 09:38.51 | bf_ | Holy moly is this all a mess. |
| 09:39.16 | bf_ | Now I have to compile dahdi from source in order for it to support the SwitchPI custom PCB |
| 09:39.53 | bf_ | Then I also need to custom-compile asterisk in order for it to support odbc connection to postgresql |
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| 12:00.23 | MarkS- | Hello, I have a question about the options around queues in Asterisk. Is it possible if a member is in 2 queues to give 1 queue always priority over the other queue? So if there are callers waiting in both queues only the priority queue is ringing and the other has to wait for that call to be answered |
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| 14:28.54 | tecfall | Why would sip headers be showing the internal ip instead of the external? I have a site that I can't connect to and i belive this is the issue. Here is the details: https://pastebin.com/PLN3c764 |
| 14:30.07 | sibiria | you're looking for externhost, not externip |
| 14:31.09 | sibiria | it can take either an IP address or a hostname |
| 14:31.23 | sibiria | also take a look at externrefresh if you intend to use a hostname |
| 14:31.26 | Samot | Or just use the right setting. |
| 14:31.44 | Samot | externaddr |
| 14:32.06 | Samot | externip=208.86.x.x <-- Not a real Chan_SIP setting. |
| 14:32.14 | sibiria | yeah externaddr if you want a static IP address |
| 14:32.22 | sibiria | externhost if you want to resolve a host regularly |
| 14:32.32 | sibiria | the latter takes an IP address as well, if needed |
| 14:32.32 | Samot | But this is why it's doing what it is. |
| 14:32.36 | sibiria | yes |
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| 18:42.40 | Someone_Else9 | any linphone users around that use the android client with push? |
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| 18:43.03 | Someone_Else9 | i'm interested in how they configured it |
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| 21:53.23 | MLC | ${CHANNEL(rtcp,txjitter)} - what are the units of measurement for this value? |
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| 22:06.45 | snuff-work | MLC: considering its 'jitter' most likely milliseconds |
| 22:07.07 | MLC | I'm getting values like 0.00416100 |
| 22:07.37 | MLC | maybe seconds? |
| 22:07.49 | snuff-work | mm.. on what 'sort' of call? ie local lan etc |
| 22:08.16 | snuff-work | if it is seconds.. then that would be 4ms |
| 22:08.21 | MLC | SIP call over the public network from a SIP provider |
| 22:08.32 | MLC | PJSIP actually |
| 22:08.54 | snuff-work | mm.. so i guess its in seconds.. as 4ms makes most 'sense' |
| 22:09.19 | snuff-work | u could also check the minrtt / maxrtt |
| 22:09.49 | MLC | that was actually my next question, all of the rtt values are 0.00000000 |
| 22:10.28 | MLC | that's a REALLY fast netowrk! |
| 22:10.47 | snuff-work | i can't remember if some of these require both ends to have a certain 'field' to work |
| 22:11.26 | MLC | I'm also at 16.2, might be better data on the most recent |
| 22:12.10 | snuff-work | well u could always just get all the possible values.. via just doing the ${CHANNEL(rtcp,all)} see what might be 'missing' |
| 22:14.17 | MLC | rxjitter and rtt are always 0.00000. Other values seem reasonable. |
| 22:18.00 | snuff-work | mm i did a test between my 2 ast.. rxjitter/rtt both 0's too |
| 22:18.20 | snuff-work | they are asterisk git sometime in last 3 months |
| 22:20.37 | MLC | any rules of thumb about jitter values. At what threshold does it become problematic to the call quality? |
| 22:21.32 | snuff-work | mm i'm used to using the MOS score.. which is 4+ great.. 3ok.. 2< bad |
| 22:22.48 | snuff-work | really if your jitter is >20ms your probably going to notice in a call.. but my recollections are hazy as i've not been deep in VOIP related stuff in ages |
| 22:25.16 | sibiria | jitter *usually* happens between your upstream trunk and the pstn |
| 22:25.42 | sibiria | and you can't measure that with qos rtp |
| 22:26.06 | MLC | right. |
| 22:26.08 | sibiria | but it's of course nice and useful to have a steady connection to your provider |
| 22:26.33 | MLC | In the current project, I'm comparing 2 providers so the qos stuff should work for that |
| 22:26.33 | sibiria | i mean, you can have issues on that first leg of the route too, obviously |
| 22:43.10 | snuff-work | mm.. interesting.. one of my 2 boxes gives rtt.. other doesn't |
| 22:43.24 | MLC | weird |
| 22:45.03 | snuff-work | mm one that gave me a rtt is running much older =) |
| 22:45.13 | MLC | lol |
| 22:45.17 | snuff-work | 2018-12-18 build.. vs 2019-08-12 |
| 22:45.30 | MLC | ghosts in the machine |
| 22:52.32 | sibiria | are you entirely sure you're receiving rtcp on the one showing 0 rtt? |
| 22:52.41 | sibiria | what does the rxcount on that machine say? |
| 22:56.46 | *** join/#asterisk Katty (uid62315@gateway/web/irccloud.com/x-jnbjzavloitqczyu) |
| 23:00.28 | snuff-work | both have rx and tx count numbers |
| 23:00.38 | snuff-work | NoOp("PJSIP/3202-00000506", "ssrc=1827324442;themssrc=3491623665;lp=0;rxjitter=0.000000;rxcount=942;txjitter=0.000038;txcount=941;rlp=0;rtt=0.000000") |
| 23:00.44 | Katty | bloop |
| 23:01.04 | snuff-work | vs NoOp("PJSIP/ast-mel-00000903", "ssrc=298585860;themssrc=3491623670;lp=0;rxjitter=0.000000;rxcount=783;txjitter=0.000249;txcount=787;rlp=0;rtt=0.013320") |
| 23:01.54 | snuff-work | oops paste the 'right' second one |
| 23:01.55 | snuff-work | NoOp("PJSIP/3202-00000506", "ssrc=1827324442;themssrc=3491623665;lp=0;rxjitter=0.000000;rxcount=942;txjitter=0.000038;txcount=941;rlp=0;rtt=0.000000") |
| 23:02.09 | snuff-work | err.. |
| 23:02.10 | snuff-work | NoOp("PJSIP/3202-00000508", "ssrc=1095312831;themssrc=3491623670;lp=0;rxjitter=0.000000;rxcount=784;txjitter=0.006077;txcount=787;rlp=0;rtt=0.000000") |
| 23:02.22 | snuff-work | copy n paste fail |
| 23:02.29 | Katty | And this is why we use pastebin! |
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