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| 01:31.02 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0 (2018/10/09), Security Only: 15.6.1 (2018/09/20); 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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| 09:36.17 | Guest9103 | Hi, we have continous issues that happens once every week approximately. The issue is that the UDP buffer fills up and asterisk stops receiving SIP. I have no idea of what is causing this at this stage. When it occured last time an hour ago I extracted some asterisk core statistics => https://pastebin.com/PFvdfVm9 |
| 09:37.02 | Guest9103 | Does this info contain any information that could be of use? the asterisk UDP buffer was stuck at 8346624 when this occured. Extracted using "netstat -peanut | grep 5060" |
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| 10:17.11 | ganapathi1 | hi . am using SIP with webrtc. |
| 10:17.42 | ganapathi1 | but my SIP channels are keep active even though operator disconnected. |
| 10:18.18 | ganapathi1 | how to destroy channels ?. it's keep active.due to some network flood from asterisk server |
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| 14:37.06 | kkocaerkek | hi all. i am working on an asterisk based predictive dialer project. i need to cancel calls immediately if called number is connected to a fax machine. how can i do that? |
| 14:41.58 | [TK]D-Fender | https://www.google.ca/search?q=asterisk+fax+detection |
| 14:41.59 | [TK]D-Fender | ^ |
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| 14:42.17 | Slade | the canadian googles |
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| 14:43.04 | Slade | i wonder if they will walk out too |
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| 15:04.25 | Guest9103 | I think AMD will be fine for this as well |
| 15:18.25 | MelMalik | gongles |
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| 15:45.45 | sibiria | asterisk's own AMD doesn't do any beep detection, by the way |
| 15:46.02 | sibiria | it only counts words |
| 15:47.07 | sibiria | also, its accuracy is quite low |
| 15:53.07 | Guest9103 | depends on what metric you are using, pickup speed and such is a great tool to get rid of fax, voicemails and IVRs. We use it with great efficiency for this |
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| 16:07.52 | sibiria | we combine word count and beep detection (our own software), achieving about 98%+ efficiency tested across millions of calls |
| 16:08.41 | sibiria | (~2% representing bad line with severe audio drop) |
| 16:11.11 | robmal | Is it for sale? |
| 16:18.58 | igcewieling | in pjsip show channelstats is the RTT correct? Most of the lines have < 1ms of RTT. https://pastebin.com/DSunGV7t |
| 16:19.42 | igcewieling | everyone of the phones are off-site from the server. |
| 16:20.56 | sibiria | robmal: it is not, no. internal use only |
| 16:21.16 | sibiria | i only know of two beep-detecting AMDs for Asterisk that are actual commercial products |
| 16:21.29 | sibiria | i would not recommend either of them after having used both extensively before we succumbed to writing our own |
| 16:22.30 | igcewieling | Asteria has a calling app with their own AMD. |
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| 20:09.18 | jrun | hello, is there a way to find out how many channels a peers allows? i.e. originating channels from us. |
| 20:16.33 | [TK]D-Fender | Who's "peers'? |
| 20:17.27 | jrun | higher tier provider |
| 20:27.29 | [TK]D-Fender | So "not your server" |
| 20:27.46 | [TK]D-Fender | tehre is no system for determining where they will cut you off short of trying & failing |
| 20:28.34 | trmg | Anyone have experience with Nortel/Avaya 1140E IP phones (SIP firmware) and Asterisk? I cannot get attended transfers to work for the life of me. Blind transfers work fine. I have Polycom SoundPoint 331/335's as well as a Cisco 7912 running SIP firmware that can do attended transfers no problem. |
| 20:28.53 | jrun | [TK]D-Fender: thanks |
| 20:29.20 | jrun | [TK]D-Fender: oh, can DEVICE_STATUS be used here? |
| 20:29.30 | [TK]D-Fender | no |
| 20:29.44 | [TK]D-Fender | That just says whether you have a channel in use against the peer or not |
| 20:37.30 | trmg | I think /buffer 7 |
| 20:37.38 | trmg | facepalms |
| 20:40.32 | MrTAP | jrun: you call them and enquire |
| 20:42.22 | jrun | MrTAP: hahaha, true. or lookup the contract. |
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| 20:57.21 | *** join/#asterisk ledoktre (~ledoktre@64-83-225-102.static.stcd.mn.charter.com) |
| 20:59.19 | ledoktre | hey greetings. quick question regarding Park command. I see the syntax on the website but am having trouble with it and am stuck. i just want to 1- silence the announcement only when manually running park in dial plan and 2- set the timeout differently than it is set in the res_parking.conf. What would be an example dialplan command to park the active channel (spot is set previously with variable)? |
| 20:59.59 | ledoktre | right now I have Park(20000,,,,s,default) - 20000 being 20 seconds, s being silence and default the selected parking lot. |
| 21:00.01 | igcewieling | Can anyone help me resolve process_text_line: Inheritance requested, but category 'endpioint-tpl' does not exist, line 17 of /etc/asterisk/pjsip.conf line 17 is the first instance of a section using a template. |
| 21:09.42 | file | did you misspell endpoint? cause that says endpioint-tpl |
| 21:10.08 | igcewieling | apparently I did. I must be tired. thanks. |
| 21:10.42 | ledoktre | In case it helps anyone, apparently the syntax has changed a bunch in a few versions. I am looking at a more recent example and it is basically flipped. thanks anyway i think i got it |
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