| 00:00.08 | bobbymc | i guess i have to use docker or some other type of linux container =( |
| 00:01.10 | bobbymc | file: thank you for verifing this, i was under the impression timefd uses software for timing, and by software i thought it doesnt need to have access tot he bus or anyhting else for timing |
| 00:01.28 | file | timerfd is the interface to the kernel for providing timing |
| 00:01.41 | file | but even with a software derived timing source that still uses the system clock in some manner |
| 00:01.49 | file | which can be variable in a virtualized environment |
| 00:02.05 | bobbymc | yea the issue is the system clock in a vm is not dependable |
| 00:02.09 | bobbymc | it falls back |
| 00:02.14 | bobbymc | thats why ntpd is a must on vms |
| 00:02.23 | bobbymc | well at least vms like xen and kvm |
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| 00:07.35 | [TK]D-Fender | heads out for the evening... |
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| 03:00.20 | priorfra | bobbymc, thank you for answering |
| 03:04.09 | priorfra | Hi all again: I'm testing video (vp8) on Asterisk 11; I have no errors, audio is ok but I cannot see video. Any idea about howto troubleshoot this issue ? I have followed all tutorials, patched, etc. Any suggestion is welcome. |
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| 08:01.12 | jzu_ | what kind of trunk settings I would need to if I have IAX2 Trunk configured and outbound route would be behind that IAX2 Trunk, is that kind of setup hard to do?-) |
| 08:01.15 | jzu_ | guess no :) |
| 08:01.48 | jzu_ | Phone -> PBX A <--IAX2--> PBX B --> SIP Trunk with outgoing DID |
| 08:02.31 | jzu_ | calls are so seldom that having just one outgoing DID that serves both PBX's would be enough |
| 08:11.13 | ChannelZ | not sure I even understand the question. You just Dial() wherever you want. If that's another Asterisk box, so be it. Then its dialplan needs to be constructed such that it Dial()s wherever it should go from there. |
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| 08:16.23 | jzu_ | ok, so yeah - that's what I meant |
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| 08:25.25 | ChannelZ | just keep in mind you're creating more latency doing that |
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| 09:52.25 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); 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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| 10:43.29 | trurl | Hi. Im' reading a book to "learn asterisk" and it says even in a voip/sip-only configuration installing DAHDI as a prereq for asterisk is recommended because of some timing modules provided. is this still valid with asterisk >13.1? (the book covers asterisk 11) |
| 10:44.53 | WIMPy | It was only required up to 1.6.something. |
| 10:47.22 | trurl | "We recommend that you install it even if you have no hardware installed, because DAHDI is a dependency required for building the timing module res_timing_dahdi and is used for Asterisk dialplan applications such as MeetMe()." |
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| 11:00.39 | file | it's *required* for chan_dahdi and app_meetme, for other stuff that requires timing there are other timing modules which do not require it |
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| 11:05.11 | angryuser_laptop | good day, what is the default value of the session-timer of the asterisk when nothing is set ? |
| 11:06.13 | trurl | file: okay, thanks, so that is still valid with * 13.1 i take. |
| 11:06.55 | file | if you can use an alternative timing module (such as res_timing_timerfd) and don't need Meetme, then it's not needed |
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| 11:36.43 | dan_j | Hi. A client is experiencing jitter on their ADSL line and sip calls to our asterisk server are barely audible. The ISP say that they cant see anything wrong and no packet loss. What options do I have apart from replacing the adsl line? |
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| 13:21.31 | malcolmd | note the app_confbridge is a nice replacement for app_meetme |
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| 13:42.48 | alex999 | hello everybody... |
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| 13:44.34 | alex999 | My asterisk server is currently at version 13. I am installing fax for asterisk following the various procedure to gather the best compiled pgm. In my case is Core2. |
| 13:45.11 | alex999 | Anyway i cannot find the core2 version 13 but just version 12. |
| 13:46.04 | alex999 | do u think I could have any problem installing core2 version 12 |
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| 13:47.36 | WIMPy | No, but I bet it will refuse to load. |
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| 13:49.37 | alex999 | is it normal that version 13 is not yet released |
| 13:50.09 | mjt | it is absolutely abnormal ;) |
| 13:50.19 | WIMPy | The guys at Digium might have other priorities. |
| 13:50.43 | WIMPy | mjt: Still using fax is perverted |
| 13:51.00 | WIMPy | Buy pretty wide spread. |
| 13:51.45 | mjt | yes indeed |
| 13:52.30 | alex999 | how can I request it... |
| 13:53.33 | alex999 | I know that few people are still using fax... but for who still does, I need to offer this service |
| 13:57.00 | [TK]D-Fender | alex999: Just use regular SpanDSP |
| 14:02.22 | alex999 | what differs from FFA & SpanDSP? Which one is the lightest? |
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| 14:05.15 | [TK]D-Fender | same thing |
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| 14:05.26 | [TK]D-Fender | From your end |
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| 14:07.47 | alex999 | D-Fender: Thank you!!! You are always very helpful! |
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| 14:10.10 | [TK]D-Fender | You're welcome |
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| 15:57.09 | cusco | hi |
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| 15:57.46 | cusco | can a sub channel, say, a dial from app Queue() to a Local/member , set a variable that can be read back in the first channel ? |
| 15:58.33 | cusco | say queue tryies: Local/member1 - returns busy(), and then Local/member2 - in this local dialplan i would like to evaluate a variable that was previously set in Local/member1 |
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| 16:33.37 | ruben23 | hi guys |
| 16:34.25 | ChannelZ | hey lady |
| 16:34.25 | ruben23 | i have added some sip configuration on the sip.conf on my asterisk but problem even i reload on the CLI> it does not take effect..i need to asterisk the service before it works |
| 16:34.35 | ruben23 | what could be the issue |
| 16:34.50 | ChannelZ | what does 'sip reload' tell you? |
| 16:35.11 | ruben23 | 206-225-84-35*CLI> sip reload |
| 16:35.11 | ruben23 | [Apr 2 12:34:58] Reloading SIP |
| 16:36.03 | ChannelZ | what is your verbose set to? |
| 16:36.40 | sgriepentrog | cusco: try https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_SHARED |
| 16:38.12 | ruben23 | ChannelZ: how to check..? |
| 16:39.01 | ruben23 | Connected to Asterisk 1.8.23.0-vici currently running on 206-225-84-35 (pid = 2767) |
| 16:39.01 | ruben23 | Verbosity is at least 26 |
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| 16:43.24 | ChannelZ | core set verbose |
| 16:43.40 | ChannelZ | grr |
| 16:43.42 | ChannelZ | core set verbose 3 |
| 16:44.59 | ChannelZ | if you're at least 3 and 'sip reload' only says that one line, then something is screwed. Check your sip.conf syntax that you didn't screw up something early on, have some wierd characters in it, etc |
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| 18:38.12 | starbuck | hi guys, is there a way to backup the configuration from a cisco converted sip phone to a tftp? |
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| 19:18.15 | jmordica | I'd like to write the number of active calls per second or minute to a file from asterisk. I'm able to "watch" the active incoming calls by running this |
| 19:18.26 | jmordica | watch "asterisk -vvvvvrx 'core show channels verbose' | grep -e SIP/bw-atl* -e SIP/bw-denver*" |
| 19:18.47 | robmal | Why not zabbix? |
| 19:18.58 | jmordica | but how would I go about writing the number of lines/calls to a file every minute or so? |
| 19:19.26 | jmordica | never heard of zabbix |
| 19:19.53 | robmal | Oh. Google zabbix/cacti/nagios/munin + asterisk |
| 19:20.04 | robmal | You'll thank me later ;-) |
| 19:21.25 | jmordica | ah cool |
| 19:21.36 | jmordica | which one would be best for monitoring/storing active calls? |
| 19:26.05 | robmal | If you need only active calls stats i'd go with cacti. |
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| 19:32.06 | jmordica | and nagios for more fine grained data? |
| 19:34.50 | robmal | Both zabbix and nagios are better suited for ensuring continiuity of service, so they can automatically react if something goes sideways (sip peer unreachable -> ping -> traceroute -> nmap -> notify), but zabbix has nicer charts ;-) |
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