| 00:11.21 | linuxfool | Hey everyone. Anybody have some examples of running asterisk on old hardware? |
| 01:09.07 | *** part/#asterisk xheliox (~jeff@pdpc/supporter/active/xheliox) |
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| 01:29.32 | ChannelZ | wonders what sort of "example" he was after |
| 01:37.52 | WIMPy | A PC? |
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| 02:06.47 | mefistofeles | so I'm trying to setup shorewall rules/policies for asterisk to work, would you happen to know a good link/tutorial for that? |
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| 02:51.07 | kim_bruning | mefistofeles, at least have port 5060 (SIP) open, and if possible ports 10000 to 20000 for RTP (check /etc/asterisk/rtp.conf to see which ports you're listening on) |
| 02:52.12 | kim_bruning | depending on who/what, you may want to restrict access to particular external IPs , or use something like openvpn to connect between phones/pbxes |
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| 03:16.25 | mefistofeles | kim_bruning: ok |
| 03:16.30 | mefistofeles | kim_bruning: thanks |
| 03:16.49 | mefistofeles | kim_bruning: this is actually a small and "simple" setup, small business with just a few extensions and such |
| 03:17.16 | kim_bruning | Ok... so you probably want to restrict connections to the outside world to just be with your upstream trunk provider or what have you |
| 03:17.29 | kim_bruning | that'll keep most of the attacks at bay ;-) |
| 03:17.46 | kim_bruning | (you'd be surprised at how many connections you get if you don't firewall) |
| 03:26.02 | mefistofeles | kim_bruning: sure, that's what I think as well |
| 04:44.16 | wyoung | hey, I am having issues with voice quality, how can I troubleshoot this? |
| 04:45.21 | wyoung | I am using ALAW codec, 15/15 Mbit Internet with about a 2 - 3ms latency, no traffic shaping or prioritising setup atm (not sure if this is the issue) |
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| 05:19.58 | jwpierce3 | I'm trying to use playback command but asterisk complains. The file exist, I've used both relative and absolute paths. I've encoded the file en both .gsm and .ul formats, same errors: |
| 05:20.18 | jwpierce3 | [Apr 12 00:11:10] WARNING[18965][C-00000000]: file.c:774 ast_openstream_full: File /var/lib/asterisk/sounds/en/bianchisgreeting.ul does not exist in any format |
| 05:20.18 | jwpierce3 | [Apr 12 00:11:10] WARNING[18965][C-00000000]: file.c:1100 ast_streamfile: Unable to open /var/lib/asterisk/sounds/en/bianchisgreeting.ul (format (ulaw)): No such file or directory |
| 05:20.18 | jwpierce3 | [Apr 12 00:11:10] WARNING[18965][C-00000000]: app_playback.c:494 playback_exec: Playback failed on SIP/flowroute-00000000 for /var/lib/asterisk/sounds/en/bianchisgreeting.ul |
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| 05:22.21 | jwpierce3 | all my devices in sip.conf have the following: allow=ulaw allow=alaw allow=gsm |
| 05:30.16 | wyoung | another issue I have having. I have copied AU sounds to /usr/share/asterisk/sounds/en-au and have specified language=en-au in sip.conf however I can still hear allison when leaving a voice message |
| 05:30.51 | wyoung | jwpierce3: don't specify the extension |
| 05:31.13 | wyoung | in your confi |
| 05:31.14 | jwpierce3 | wyoung, I've tried it both ways |
| 05:31.14 | wyoung | g |
| 05:31.33 | wyoung | jwpierce3: your configuration has no extention while the physical file needs to have one |
| 05:31.56 | jwpierce3 | at this point I'm trying to call into the pbx from the outside, so I'm not really touching the extensions |
| 05:32.13 | wyoung | file extensions |
| 05:32.17 | wyoung | filename.ext |
| 05:32.23 | wyoung | .ext is the file extention |
| 05:32.31 | jwpierce3 | ah |
| 05:33.09 | wyoung | so instead of having bianchisgreeting.ul in your config put in bianchisgreeting |
| 05:33.40 | wyoung | I am not sure is ul as the extension will work, I am not sure how asterisk requires it to have the alaw or ulaw or wav extension |
| 05:34.24 | jwpierce3 | wyoung, well, that worked |
| 05:34.32 | jwpierce3 | what a pain in the a&& |
| 05:34.32 | wyoung | :D |
| 05:34.35 | wyoung | \o/ |
| 05:34.37 | wyoung | \o\ |
| 05:34.40 | wyoung | <PROTECTED> |
| 05:35.17 | jwpierce3 | thank you sir |
| 05:35.23 | wyoung | any time buddy |
| 05:35.28 | wyoung | now, solve my problem! :) |
| 05:37.40 | jwpierce3 | wish I knew more to help |
| 05:42.00 | wyoung | If I set(CHANNEL(language)=en-au) it works, however it is ignoring my settings in sip.conf (and the call is coming in from sip) |
| 05:42.18 | wyoung | do I need to set the language=en-au in extensions.conf too? |
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| 07:55.31 | shtrb | I'm having trouble understaing a case (client is asterisk I'm not) , a prack is being sent but with a different branch (different from the original invite) and as I understand it should have the same branch is that behavoir nromal ? |
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| 08:57.49 | wyoung | shtrb: no idea |
| 08:59.09 | shtrb | wyoung, thank you |
| 08:59.38 | wyoung | 3any time! |
| 08:59.55 | shtrb | :) |
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| 09:49.48 | wyoung | so, any one on? |
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| 09:53.51 | bounceman | Hi, do you know of any parser for /var/log/asterisk/messages? It's kinda impossible to read and get sensible data out of when you reach higher call volumes with queues, agi, pauses and such. |
| 09:56.31 | WIMPy | grep |
| 09:56.58 | shtrb | sed ? |
| 09:57.17 | Zogot | what does round robin do in asterisk queues, we have 8 phones and we seem to just see rrmemory and rrordered coming up as the same |
| 09:58.38 | bounceman | WIMPy: grep? I'd like to see those arguments, sounds like one hefty piece of art |
| 09:59.18 | bounceman | Sure I can grep every line where the agent name occurs, but that will exclude all surrounding information related to that event that does not include the agent name |
| 09:59.24 | WIMPy | The call id? |
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| 10:00.22 | bounceman | Callid will only work with chan_sip not pbx.c and reg.agi |
| 10:00.34 | bounceman | I need something more cleaver |
| 10:03.36 | WIMPy | Guess I'm not using any features, where it fails. |
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| 10:16.36 | wyoung | yay WIMPy! |
| 10:17.55 | wyoung | WIMPy: I set language=en-au in sip.conf however when I call my voicemail macro it changes back to language=en. I have to explicitly set CHANNEL(language)=en-au within the voicemail macro for it to work. |
| 10:18.17 | wyoung | do macros get their default language from another place? or do I need to specify a language in extensions.conf? |
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| 10:33.57 | WIMPy | asterisk.con > the channel > the dialplan, AFAIK. |
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| 11:03.37 | wyoung | hmmmm |
| 11:04.51 | wyoung | if I set the default language and it doesn't contain all files that are in alisons / en-US's sounds directory will it blow up? |
| 11:06.37 | wyoung | blow up / not fallback to en-US and log an error? |
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| 11:21.45 | wyoung | so, how does one force SSL/TLS for SIP and RTP? |
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| 11:24.21 | jonah | Hi does any body use an x100p card or know anyone that does so I can please copy the config files? I can't get mine working, thanks for any help at all |
| 11:25.43 | wyoung | I use digium all the way! |
| 11:25.48 | wyoung | pew pew! |
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| 11:48.28 | phpboy | hi everybody, I'm trying to compile dahdi in CentOS 7 but it doesn't seem to want to build. getting unsigned driver around usb checks. |
| 11:48.46 | phpboy | anybody have any experience with dahdi on CentOS 7? |
| 11:49.36 | phpboy | The reason why I'm building dahdi is because my conference calls without dahdi is not working. CentOS 7 on vmware... I'm thinking it's a timing issue. Conference calls are often choppy (using meetme) |
| 11:51.19 | Gugge | phpboy: dont use meetme then :) |
| 11:52.10 | phpboy | Gugge: what else can I use for conferencing? wasn't aware of anything else |
| 11:52.38 | Gugge | Conference() |
| 11:52.42 | Gugge | meetme() is deprecated |
| 11:54.55 | Gugge | sorry, confbridge |
| 11:54.59 | Gugge | not conference :) |
| 11:55.51 | phpboy | I'll be damned |
| 11:56.01 | phpboy | I am using app_confbridge |
| 11:56.13 | phpboy | I should probably try meetme then lol |
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| 11:57.12 | Gugge | that timing module are you using? |
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| 12:17.25 | pawiecki | Hello! Do you know how to simply check last registration time for a given sip account? Let's say "sip show peer x" and /var/log/asterisk/messages doesn't give a clue. |
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| 12:26.32 | wyoung | does asterisk support sip requests from snom phones to start recording active channel? |
| 12:27.06 | wyoung | snom docs: "The PBX has to support INFO-Requests and has to understand the Record:-Header" |
| 12:39.30 | file | yes. |
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| 12:41.53 | ruben23 | hi guys anyone cna help somehow for this error compiling asterisk 11 -----> http://pastebin.com/GnTkiFej |
| 12:43.08 | [TK]D-Fender | "Either run 'make menuselect' or remove the existing |
| 12:43.09 | [TK]D-Fender | <PROTECTED> |
| 12:43.23 | [TK]D-Fender | Was there a problem following what it told you very directly to do? |
| 12:45.37 | ruben23 | <PROTECTED> |
| 12:45.54 | ruben23 | if i seletc it on the menu select..? there are so many modules.. |
| 12:45.55 | [TK]D-Fender | ? |
| 12:46.22 | [TK]D-Fender | It tells you 2 things. |
| 12:46.29 | [TK]D-Fender | Go in there and try both if you have to |
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| 12:46.40 | ruben23 | TK]D-Fender: i mean what modules i should enable or disable..? |
| 12:46.51 | [TK]D-Fender | JUST RUN IT. |
| 12:47.29 | [TK]D-Fender | "or remove the existing menuselect.makeopts file to resolve this issue." |
| 12:47.41 | pawiecki | Shia LaBeouf© |
| 12:47.56 | ruben23 | [TK]D-Fender: sorry im not sure this is right ---> menuselect.makeopts: command not found |
| 12:48.16 | [TK]D-Fender | That is a FILE not a COMMAND |
| 12:48.45 | ruben23 | ok, im on the make menuselect now |
| 12:49.11 | ruben23 | there are so many to select somehow... |
| 12:49.27 | ruben23 | should it be ---> Compiler Flags |
| 12:49.53 | ruben23 | XXX BUILD_NATIVE <-------- is already selected and i cant removed it |
| 12:50.20 | [TK]D-Fender | Did it tell you to REMOVE anything in tthat notice? |
| 12:50.37 | ruben23 | TK]D-Fender: sorry no. |
| 12:51.22 | ruben23 | TK]D-Fender: what to do somehow here..? |
| 12:51.25 | [TK]D-Fender | so finish menuselect saving your changes and recompile. |
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| 13:59.50 | bertrand_ | Hello folks, just installed asterisk / dahdi on Debian 8.4 and can't find any dahdi init script. Is that related to systemd switch? |
| 14:08.22 | newtonr | bertrand_, you ran "make config" ? |
| 14:10.24 | *** join/#asterisk seik0 (~seik0@5.142.56.179) |
| 14:10.44 | bertrand_ | installed the debian packages : dahdi 1:2.10.0.1-1 , dahdi-linux 1:2.10.0.1~dfsg-1 , dahdi-firmware-nonfree 2.10.0-1 , dahdi-source 1:2.10.0.1~dfsg-1 and compiled dahdi-modules-3.16.0-4-amd64:amd64 1:2.10.0.1~dfsg-1+3.16.7-ckt25-2 using module-assistant |
| 14:11.20 | bertrand_ | none of these packages seems to contain init.d scripts nor systemd stuff, but i'm not really fluent with systemd, yet |
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| 14:12.02 | seik0 | Hi, guys. Can't find solution to situation: I make originate and create ChanA, then I redirect ChanA to dialplan where I make Dial , so I create ChanB. Then I want to redirect ChanB (and detach ChanA), but I can't get ${CHANNEL} of ChanB |
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| 14:18.19 | [TK]D-Fender | When do you want to do this? What is making this decision? |
| 14:19.22 | seik0 | ChanA is operator, ChanB is client, operator needs to pass client further |
| 14:20.17 | seik0 | and why first to call operator? in order to client not to wait anything |
| 14:20.48 | [TK]D-Fender | ? |
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| 14:21.21 | seik0 | to imitate situation when operator dials client number manually |
| 14:21.27 | [TK]D-Fender | Your wording is confusing. Please break this down. |
| 14:21.40 | seik0 | mm |
| 14:22.03 | [TK]D-Fender | Who triggers the originate? When "A" answers the call they are sent to the dialplan calling DIAL(). |
| 14:22.31 | [TK]D-Fender | That call will eventually get answered. Att what point do you want "B" to get sent somewhere else? |
| 14:22.44 | [TK]D-Fender | What triggers that decision to start the redirect? |
| 14:22.52 | [TK]D-Fender | Is that supposed to be immediately upon answer? |
| 14:23.00 | [TK]D-Fender | Or after they've spoken? |
| 14:23.35 | seik0 | ClientCenterMegaAutomation system decides, that specific type of connection should be initiated and originates call to operator (this creates ChanA) |
| 14:23.57 | [TK]D-Fender | Does the operator know that Orignate is coming at them? |
| 14:24.11 | seik0 | yes, he knows |
| 14:24.20 | [TK]D-Fender | as in : did they request it? Or is this just a regular automated thing? |
| 14:24.23 | [TK]D-Fender | Ok, good. |
| 14:24.45 | [TK]D-Fender | So "A" answers, then "B" gets called |
| 14:24.56 | seik0 | yes |
| 14:25.04 | seik0 | then they talk |
| 14:25.05 | [TK]D-Fender | "B" will then answer. When do you want "B" to get sent somewhere else? |
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| 14:26.14 | seik0 | after some talk operator decided, for example, to move client to VM, or to some MusicOnHold and InformationOnHold loop |
| 14:27.04 | seik0 | at low level it means to move ChanB to dialplan |
| 14:27.25 | seik0 | detaching ChanA |
| 14:28.12 | [TK]D-Fender | Why doesn't the operator just do a transfer on their phone? |
| 14:28.38 | [TK]D-Fender | This doesn't sound like anything that requires Asterisk to have any special configuration.... |
| 14:29.52 | seik0 | operator just presses button in its desktop application |
| 14:30.46 | [TK]D-Fender | What do you expect to have happen to "A"? |
| 14:30.52 | seik0 | Hangup |
| 14:31.15 | [TK]D-Fender | Ok, then use an AMI Redirect on "B" to send them where you want them to go in the dialplan to continue oneward |
| 14:31.18 | [TK]D-Fender | onward* |
| 14:31.42 | seik0 | ok, but I need to know "B" ${CHANNEL} |
| 14:32.36 | [TK]D-Fender | Look at "A" and you will see the BridgeChan |
| 14:32.43 | [TK]D-Fender | that's how you can get it |
| 14:33.07 | seik0 | cool! thanks! |
| 14:33.27 | [TK]D-Fender | or in your Dial() you can pass "A"'s channelname into a MACRO called against "B" when they answer. |
| 14:33.57 | [TK]D-Fender | Or another unique ID type variable you set in "A" using your Originate |
| 14:34.12 | [TK]D-Fender | "core show applciation dial" <- |
| 14:34.15 | seik0 | it was another option I was checking while talking |
| 14:34.16 | [TK]D-Fender | Macro/gosub |
| 14:34.41 | seik0 | I already use that Macro, but was not sure if ${CHANNEL} is accessible |
| 14:34.56 | seik0 | because Macros are specific on variables |
| 14:35.24 | seik0 | but you suggest not to use, but pass it |
| 14:35.30 | [TK]D-Fender | You can pass "A"'s channel as a parameter... and set another var in "B" to hold it. |
| 14:35.52 | [TK]D-Fender | pass it so ARG1 = ${CHANNEL} |
| 14:35.56 | [TK]D-Fender | etc |
| 14:35.58 | seik0 | I understand |
| 14:36.15 | seik0 | with "A" ${CHANNEL} everything is clear |
| 14:36.21 | seik0 | why I need to pass it? |
| 14:37.09 | [TK]D-Fender | To help you identify "B" so you know what channel to Redirect |
| 14:39.02 | seik0 | looks like ${CHANNEL} in dial-Macro is already of "B" |
| 14:39.55 | [TK]D-Fender | TThat's why I told you to PASS it as an arg |
| 14:41.03 | seik0 | ok, I understand |
| 14:41.29 | seik0 | just was thinking in terms of my application, where I have more information |
| 14:41.57 | [TK]D-Fender | yes and you can push that information into the channels to make them easier to identify so you can toss them around using AMI, etc |
| 14:42.25 | [TK]D-Fender | Originate lets you set variabels in "A". yoru Dial() can pass those values on to "B" in an easy way making "B" easily trackable |
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| 14:49.59 | seik0 | thanks |
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| 15:40.47 | seik0 | when I execute Macro in Dial, does ${UNQIUEID} inside macro is the same as of callee? |
| 15:41.15 | seik0 | because it is not or I am missing something |
| 15:41.35 | [TK]D-Fender | no, it is a separate call. |
| 15:41.40 | [TK]D-Fender | they just happen to be bridged |
| 15:42.30 | seik0 | ok then, thanks! |
| 15:50.32 | *** join/#asterisk Mango45 (~Mango45@d142-59-245-22.abhsia.telus.net) |
| 15:51.55 | Mango45 | I would like to set up a BLF key on my phone that is lit when a specific SPYGROUP is set. Can anyone suggest how I would do this? |
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| 15:53.59 | [TK]D-Fender | Define "set". |
| 15:54.43 | Mango45 | If I am able to use Chanspy to listen to something, the light should be on. |
| 16:01.46 | lvlinux | Hey guys. I'm about to setup a pfSense firewall in front of an Asterisk server. I've read that some people use a SIP proxy (siproxd) on the pfSense box. Is that something I should look further into, or just do the regular Asterisk NAT settings? |
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| 16:09.01 | Mango45 | I think I found it. Looks like func_devstate will do what I want. |
| 16:09.14 | Mango45 | module load func_devstate |
| 16:09.16 | Mango45 | whoops. |
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| 16:18.39 | [TK]D-Fender | Mango45, When would the value be set in the first palce? |
| 16:18.42 | [TK]D-Fender | place* |
| 16:19.04 | [TK]D-Fender | lvlinux, Keep your firewall OUT of this |
| 16:19.36 | lvlinux | [TK]D-Fender: so just use the regular NAT settings and forward the RTP ports? |
| 16:19.42 | [TK]D-Fender | yes |
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| 16:21.20 | lvlinux | K will do. That's what I originally planned but then saw some talk about using siproxd with pfSense, and wondered if it "solved" any problems. But what I read didn't really tell WHY they used it (probably they didn't know how to configure * NAT I guess :-) |
| 16:21.34 | Mango45 | [TK]D-Fender, it is set by the dial plan when the user hits a specific IVR. |
| 16:24.23 | [TK]D-Fender | Mango45, Then I'd set a GROUP() function against the call and a DEVSTATE against it |
| 16:26.37 | Mango45 | Cheers! |
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| 16:57.59 | dasjoe | Is it unwise to put asterisk in a container like lxc? |
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| 17:17.38 | Mango45 | HAEHAHEHA IT WORKS!!! |
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| 17:21.24 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 17:34.53 | mub | dasjoe: I've done it, but never used it in production |
| 17:35.35 | mub | I say go for it! |
| 17:47.16 | lvlinux | dasjoe: I don't have experience with LXC but I have an Asterisk production server running on OpenVZ, which is similar. It's not under a high load, but it works perfectly. |
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| 17:48.51 | [TK]D-Fender | https://www.google.ca/#q=asterisk+under+lxc |
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| 18:04.14 | drab | hi, been googling a bunch but can't find how to disable call screening with follow me (please press 1 to accept, etc). Is there a way to do that or should I simply stop using followme and implement calling a mobile from the dialplan if the extension doesn't pick up? |
| 18:25.54 | rmudgett | drab: You cannot disable the followme screening so you'll have to implement it in dialplan. |
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| 18:47.14 | dasjoe | mub, lvlinux, thanks! I'll build a small container and we'll see how it goes |
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| 19:46.02 | xheliox | Simple and probably dumb question. What lib do I need to have installed so that Asterisk builds with color support? |
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| 20:58.12 | gusto | I have a strange problem with my asterisk here installed on a mips64 cavium and running openbsd on it, that when the cpu comes under load, or just when i like activate rtcp debug asterisk (13) shuts itself off, with the error code "signal 11" but there is no dump or something |
| 20:58.59 | lvlinux | gusto: latest version of Asterisk 13 I presume? |
| 20:59.08 | gusto | sofar this strange behaviour I only noticed to Asterisk, yes, Asterisk 13 |
| 20:59.19 | gusto | there are no other packages of it in the ports of openbsd |
| 20:59.25 | gusto | for mips64 at least |
| 20:59.51 | lvlinux | Oh so it's a packaged build? |
| 21:00.31 | gusto | yes |
| 21:00.54 | gusto | and I also seen Signal 4 and Signal 5, but most of the time it is Signal 11 |
| 21:01.09 | lvlinux | ah, ok. Well you might try a manual compile and see if it makes any diff. |
| 21:01.44 | gusto | why do you think would it help? |
| 21:02.39 | lvlinux | Well, for one thing, you'd have control over exactly the source version you were getting. May have been a patch applied that fixes your problem recently. |
| 21:02.44 | gusto | and what do you think I should compile from, the sources by myself, or just checking out the openbsd ports from cvs and compile that |
| 21:04.55 | lvlinux | the sources yourself |
| 21:06.32 | gusto | so not going over the ports |
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| 21:12.04 | lvlinux | I'd give that a shot. It's the first easy step before you get into debugging junk... |
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| 21:49.04 | gusto | ok, but I try first compiling their new version, because they have in the current cvs the 13.8, so that would be very close to current, and when that does not change things, I ll compile a git HEAD |
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| 22:55.23 | diegoviola | hi |
| 22:55.35 | diegoviola | I have some credit in a SIP account that I need to deplete, where can I call for that? |
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| 23:37.31 | FuriousGeorge | hey all |
| 23:38.27 | FuriousGeorge | anyone using centos 7 / asterisk 13? it seems that the systemd script does not work because it assumes the pid should be in /var/run/ but centos and other distros mount this as tempfs |
| 23:39.17 | FuriousGeorge | so you make install the script, reboot, and find that * won't start. has anyone run into that? |
| 23:40.13 | [TK]D-Fender | that doesn't add up |
| 23:40.18 | [TK]D-Fender | tempfs = fs |
| 23:40.23 | [TK]D-Fender | that folder ... is a folder |
| 23:40.39 | [TK]D-Fender | and asterisk.conf tells * where to look for the PID. Fix it |
| 23:40.58 | [TK]D-Fender | And the startup scrip tells * where the base config folder is |
| 23:43.36 | FuriousGeorge | let me see the startup script |
| 23:47.29 | FuriousGeorge | [TK]D-Fender: i can see the pid isn't there |
| 23:47.34 | FuriousGeorge | and i know the script worked |
| 23:47.45 | FuriousGeorge | i made a pid file and rebooted and it's gone |
| 23:48.15 | FuriousGeorge | so my assumption is the systemd script does not know to make the pid file., im looking for it now |
| 23:52.06 | FuriousGeorge | im totally wrong. must have misread something |
| 23:55.24 | RadicalDev | Is it possible to get asterisk to authenticate based on something other than the fromuser? |
| 23:55.33 | diegoviola | does asterisk supports pjsip now? |
| 23:57.36 | RadicalDev | example of my peer context: http://pastebin.com/raw/5qaFqHFU |
| 23:58.02 | RadicalDev | if I sent INVITE from kamailio, asterisks ends 401, then kamailio sends the auth headers |
| 23:58.25 | RadicalDev | if fromuser is inbound, it works. If it's anything else, I get "no matching peer '1234'" |
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| 23:59.26 | RadicalDev | ideally, the authorization username, realm, and password would be enough to figure out the inbound context should be used |