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| 03:36.01 | dijib | Serious question guys does asterisk have a limit to the number of voicemail users are checking vm simultaneously? |
| 03:36.43 | WIMPy | I can't see where that should come from. Unless you are talking about the same mailbox. |
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| 03:38.10 | dijib | nvmd |
| 03:38.26 | dijib | just a weird question someone had, i couldnt find the answer |
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| 04:00.43 | dijib | so okay does anyone know of an example asterisk.service file? |
| 04:07.14 | leftleg | hello all |
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| 04:19.34 | leftleg | i have 3 asterisk boxes sip trunked. as of a few days ago I cannot dial into the conference bridges on the main pbx. i can still dial the extensions on the main pbc form the other two.. any ideas? |
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| 04:48.52 | xochilpili | hi all, i was trying to make a trunk between an avaya IPO 500v2 and asterisk with asteriskNOW distro (current), but i never could make it connect each other, does anyone has some experience with this? |
| 04:52.37 | janicez | mew |
| 04:52.57 | janicez | leftleg: idk,..? |
| 04:53.51 | WIMPy | leftleg: Check your dialplans. |
| 04:54.13 | WIMPy | xochilpili: SIP doesn't connect. What do you have? |
| 04:54.27 | xochilpili | WIMPy, Hey! |
| 04:56.33 | xochilpili | i have a freepbx asterisknow and so on, now, in the avaya, everyone in "google research" said, that i can make a trunk to connect asterisk and avaya, to call each other by extensions... i have 3 phones as i can research, they are h323 protocol. So i cant make them connect just like sip to asterisk directly, that's why i need avaya (as i can understand) |
| 04:57.27 | WIMPy | You can use H.323 with Asterisk as well. |
| 04:58.00 | WIMPy | If you need help with configuring FreePBX, you have to ask in #freepbx. |
| 04:58.16 | xochilpili | WIMPy, yes, i load the module ooh323 and make the ooh323.conf but i there im lost, between the ooh323 and phones... |
| 04:58.49 | xochilpili | tk-defender told me something that i cant forget; "dont mix up things" (or something like that) |
| 04:59.03 | WIMPy | Good plan. |
| 04:59.31 | xochilpili | one thing, the autoprovisioning of the phones, and the other side; contact with asterisk an users with their extensions |
| 05:00.00 | WIMPy | But when you use FreePBX, you need to find out how to do things its way instead of writing config files. They will usually get overwritten soon, unless you know exactely what's going on. |
| 05:00.39 | WIMPy | I'd leave provisioning for the next stage. |
| 05:00.52 | xochilpili | WIMPy, yeah, well... |
| 05:01.11 | xochilpili | schema 1: ASterisk -> h323 -> phones |
| 05:01.22 | xochilpili | schema 2: ASterisk -> trunk -> Avaya -> Phones |
| 05:01.58 | WIMPy | Well, "trunk" doesn't really exist in VOIP, other than in IAX. |
| 05:01.59 | xochilpili | schema 2 is with most info i found in google. |
| 05:02.52 | xochilpili | WIMPy, are you trying to confuse me? :P |
| 05:03.34 | WIMPy | Whoever uses the term "trunk" confuses. |
| 05:04.29 | WIMPy | Best not to use it. Just like "DID". |
| 05:04.53 | xochilpili | WIMPy, ah, well, as i can understand, the "trunk" will be something like a bridge between asterisk and avaya, am i correct? |
| 05:06.00 | WIMPy | That's not the way things work. You can create an account on one side and use it from the other. |
| 05:06.19 | WIMPy | Or in both directions. |
| 05:06.48 | WIMPy | It can involve registering or you can configure hosts/IPs. |
| 05:08.00 | xochilpili | by account you meant an extension with user to be auth? |
| 05:08.41 | WIMPy | No. An extension is somethign you dial. An account is the username/password you need to place a call or to register a peer. |
| 05:09.31 | xochilpili | in that case, where am i supposed to create that account? Asterisk or Avaya? |
| 05:10.03 | WIMPy | On both. |
| 05:10.15 | xochilpili | jeje |
| 05:10.17 | xochilpili | what?! |
| 05:10.19 | xochilpili | :P |
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| 05:10.30 | xochilpili | the same account in both sides? |
| 05:10.41 | WIMPy | This is P2P, not client/server. Unless you want to use registrations. |
| 05:10.58 | WIMPy | But only for that detail. |
| 05:12.22 | xochilpili | again, please. I have to create the same account in both places? |
| 05:12.48 | WIMPy | Yes, they need to match. |
| 05:14.48 | xochilpili | In anyplace of this; freepbx or IPO v2 manager tells any of ips when i create an account |
| 05:15.12 | xochilpili | where am i supposed to declare who is connect with? |
| 05:15.43 | WIMPy | There is no "connection". |
| 05:16.02 | WIMPy | Either you specify the peers hostname/IP or you register one to the other. |
| 05:16.07 | WIMPy | Two different concepts. |
| 05:18.31 | xochilpili | wait, im going to reset this thing (avaya) and start over... |
| 05:18.40 | xochilpili | brb |
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| 05:36.31 | xochilpili | WIMPy, still here? |
| 05:37.22 | WIMPy | yes |
| 05:38.51 | xochilpili | i reset the avaya to start over |
| 05:39.37 | xochilpili | and i create a vlan in the switch only for voice, in there i have just the asterisk and the avaya, and no internet |
| 05:42.16 | xochilpili | the thing that is not plenty clear is; once created the same account in both sides, where should be the registration settings? |
| 05:44.03 | WIMPy | 1st of all, how are they going to talk to each other? SIP or will it be H.323 as well? |
| 05:44.24 | WIMPy | Because I assumed that part would be SIP. |
| 05:45.29 | xochilpili | well, lets asume they will be SIP |
| 05:45.48 | WIMPy | ok |
| 05:46.11 | xochilpili | as i read in the manual of this avaya, the phones should be h323 and the avaya supports SIP |
| 05:46.54 | janicez | lol wat |
| 05:47.01 | xochilpili | so, i create an user in avaya; called avaya, with no extension |
| 05:47.41 | WIMPy | has no idea what that thing might support. |
| 05:48.58 | WIMPy | But I'm sure it would make a difference between an account used for a phone and one used as an uplink ("trunk"). |
| 05:51.48 | xochilpili | well, where should be created the user in freepbx? add user? |
| 05:52.08 | WIMPy | #freepbx |
| 05:54.43 | xochilpili | well, it's done... |
| 05:57.34 | WIMPy | First call made? |
| 05:57.44 | xochilpili | WIMPy, i create the same user in both sides, but i where should be registered? |
| 05:57.57 | xochilpili | no, there's no extension |
| 05:58.37 | WIMPy | You always have to specify a host one one side. Then either you register to the other or specify a host there as well. |
| 05:58.38 | xochilpili | should i create a "trunk" in freepbx? |
| 05:59.48 | xochilpili | WIMPy, when i create that user, there's no option to specify a host (in both sides) |
| 05:59.51 | WIMPy | I don't think so, but I don't know much about it. Ask in #freepbx. |
| 06:00.20 | WIMPy | If you can't specify a host , that would imply that the other end has to register. |
| 06:00.51 | WIMPy | Or that you can only receive calls via that account. |
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| 06:18.38 | xochilpili | WIMPy, i can't make it |
| 06:20.15 | WIMPy | Well, I'll try some other annoying thing: Get some sleep. |
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| 06:36.54 | xochilpili | WIMPy, im lost |
| 06:37.52 | xochilpili | i created a user, and two extensions, in the avaya there's a part to make a "Line" that i can put a ip/host but i dont receive anything in CLi |
| 06:49.05 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 06:49.37 | _abc_ | Hello. Asterisk question: Is there a way to limit incoming and or outgoing simultanous connections through a sip trunk or iax trunk without scripting? |
| 06:50.05 | _abc_ | I could not find anything of the kind, I would like to limit simultanous calls through a vpn between 2 *'s. |
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| 07:50.28 | ChannelZ | _abc_ you can mess with callcounter/busylevel in sip.conf |
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| 07:52.35 | _abc_ | ChannelZ: Not at dialplan level? |
| 07:52.43 | ChannelZ | for IAX I dunno. Generally you implement in the dialplan. |
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| 07:54.47 | ChannelZ | see function GROUP, GROUP_COUNT... |
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| 08:09.35 | _abc_ | ChannelZ: thanks that was what I was looking for, I think |
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| 08:22.10 | Fira | Hey there, I'd need a hand with generic multi-PBX architecture questions... Let's assume i have a bunch of Asterisks across the place, and want to be able to phone from one to another. They can't neccessarily be fully meshed. How to achieve that ? Is it possible to route calls through multiple SIP Trunks, or do you need some kind of softswitches to handle that ? |
| 08:23.05 | Fira | I've been searching for examples but most of the Trunking scenarios I can find on the Internet are simple trunking to a SIP Provider :/ |
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| 08:28.24 | ChannelZ | It's not really any different. |
| 08:28.38 | ChannelZ | You make a peer on each side and dial one from the other. |
| 08:29.02 | ChannelZ | How you write the dialplan to make sense is up to you |
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| 08:32.08 | Fira | Oh great :) That's what I hoped... Thanks, i'll experiment with em'. Sadly that'd be to run a home-made ARI application so I fear it might take some extra work to properly adapt it... But that should fun :D. Thanks a lot. |
| 08:32.41 | ChannelZ | Sure good luck |
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| 09:23.45 | _abc_ | How come everyone asks about trunking 10 hours after I asked? Haven't seen a trunking question here in days, or weeks. |
| 09:23.51 | _abc_ | Excepting my own |
| 09:24.23 | _abc_ | <PROTECTED> |
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| 10:16.59 | wonderworld | Calling mom sounds best with the familiar G.711 encoded representation of her voice. |
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| 12:15.36 | Rico | hello |
| 12:15.52 | Rico | does chan_dahdi has a relation with system clocksource ? |
| 12:15.59 | Rico | (tsc, hpet, ...) |
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| 13:01.09 | Rico | where do I find mysql schemas for asterisk 13.4 ? nothing about sip peers in contrib/realtime/mysql/ |
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| 13:09.58 | Rico | need help about that please |
| 13:10.02 | Rico | does chan_dahdi has a relation with system clocksource ? |
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| 13:25.18 | Rico | does chan_dahdi has a relation with system clocksource ?/j #centos |
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| 14:18.06 | newtonr | Rico, https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces |
| 14:18.46 | newtonr | "As long as the dahdi module is loaded, it will provide timing to Asterisk either through installed telephony hardware or utilizing the kernel timing facilities when separate hardware is not available." |
| 14:21.29 | Rico | newtonr: ok thanks |
| 14:22.04 | Rico | mi kernel timer is "jiffies", it seems to be the worst one |
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| 14:33.02 | Rico | any idea about mysql schemas for asterisk 13 ? |
| 14:33.07 | Rico | where can I find it ? |
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| 14:56.42 | newtonr | Rico, https://wiki.asterisk.org/wiki/display/AST/Managing+Realtime+Databases+with+Alembic |
| 15:01.51 | Rico | newtonr: thanks |
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| 15:14.06 | russlevy | [TK]D-Fender: you suggested a couple days ago to "Go call an external script in M() off your dial " -- you mean macro right? |
| 15:17.16 | WIMPy | Macros are deprecated. Use Gosub if you write something new. |
| 15:18.14 | russlevy | WIMPy: trying to implement monitoring of an opus channel at a higher khz, and monitor only does it at the bridged rate, 8khz |
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| 15:20.08 | WIMPy | Oh, that was to trigger someting to start the monitor via AMI, right. |
| 15:20.24 | russlevy | should i theoretically be able to do it through ARI in python/node? |
| 15:20.30 | WIMPy | Maybe you already got enough information without doing anything? |
| 15:21.11 | russlevy | WIMPy: what do you mean? |
| 15:21.16 | WIMPy | I think the bridge event would occur after a Macro or Gosub finishes anyway. |
| 15:21.47 | WIMPy | Via AMI you get very detailed information about what's going on anyway. |
| 15:21.57 | russlevy | would ARI be the way to go? and create the bridge there? i.e. combine https://wiki.asterisk.org/wiki/display/AST/ARI+and+Bridges%3A+Basic+Mixing+Bridges with https://wiki.asterisk.org/wiki/display/AST/ARI+and+Media%3A+Part+1+-+Recording or something |
| 15:22.05 | russlevy | (make my own bridge, as well as make my own recordings) |
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| 15:22.42 | WIMPy | Actually you wouldn't have to wait for the bridge ass long as you already know the name of the channel, so that might be ok. |
| 15:22.56 | WIMPy | Yes, you could do it that way as well. |
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| 15:26.34 | russlevy | WIMPy: last question: via ari (python) can i control a confbridge? |
| 15:27.20 | WIMPy | I have not looked in to ARI at all. I find the interface too ugly. |
| 15:27.32 | russlevy | ok :) thanks |
| 15:28.41 | russlevy | it seems overly complicated, but looks miles better than AGI and dialplan macros (which I used years ago) |
| 15:28.59 | WIMPy | Well, that's the idea. |
| 15:29.09 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 15:29.32 | WIMPy | Everything you can do via AGI or AMI and a little more in one. |
| 15:30.05 | russlevy | yah -- maybe just need a v2 cleanup with a nicer interface |
| 15:30.30 | WIMPy | But juggling with multiple sockes doesn't look reliable to me. |
| 15:31.27 | iulhk | Through AGI, best way to check sip peer status, online or offline, i am getting data from sippeers.lastms from sippeers table, is there a better way to check either sip peer is online or offline ? |
| 15:32.30 | WIMPy | status |
| 15:32.53 | russlevy | thanks WIMPy -- have a great weekend |
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| 15:35.22 | Resnik | is asterisk 13 ok to try/use? |
| 15:38.25 | hatlessman | I've been using 13 in production for a month or so. Its pretty stable. |
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| 15:57.36 | Resnik | is there a way to check if there are any changes in specific app between 12 and 13 asterisk version? |
| 16:04.41 | malcolmd | the UPGRADE[-x].txt files in the tarball should describe the changes. you can peruse them to see if there's a change to an application you're curious about |
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| 17:37.30 | hatlessman | Has anyone actually used chan_pjsip and websockets? |
| 17:37.56 | hatlessman | So far its been a nightmare for me. Segfaults, DTLS setup failures |
| 17:38.25 | hatlessman | WS + chan_sip works great. |
| 17:42.35 | dijib | does anyone have an example of a working systemd script to start asterisk13 on a 2015 arch |
| 17:42.46 | dijib | ? |
| 17:44.20 | ChannelZ | is dreading systemd |
| 17:53.31 | ChannelZ | And yes on pjsip, no on websockets |
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| 19:02.36 | hatlessman | dijib: https://aur.archlinux.org/packages/as/asterisk/asterisk.tar.gz |
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| 20:09.04 | K0HAX | Is there a reason to use pjsip over sip? |
| 20:13.05 | superscrat | K0HAX, here is an old blog post from mjordan: http://blogs.digium.com/2013/11/20/asterisk-12-part-iv-sip-stack-future/ |
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| 20:25.47 | xochilpili | hi all |
| 20:25.59 | superscrat | howdy, xochilpili |
| 20:26.18 | xochilpili | can anyone give me a hand with asterisk and avaya IPO 500 v2? |
| 20:26.46 | xochilpili | i can't make them connect each other... |
| 20:31.58 | [TK]D-Fender | heads home |
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| 20:45.39 | hatlessman | K0HAX: PJSIP has 100rel |
| 20:46.20 | hatlessman | 100rel allows ICE to start early on webrtc calls. |
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| 23:44.28 | xochilpili | hi all |
| 23:45.07 | xochilpili | i have created a trunk from asterisk to an avaya IPO 500 V2, i followed this guide : http://www.mehrdust.com/archives/asterisk-tips-for-integration-with-avaya |
| 23:46.04 | xochilpili | and all the outputs in there, i have them, i create 2 extensions one: 1001 in asterisk with freepbx and 8000 in avaya |
| 23:46.39 | xochilpili | i have an avaya h323 phone connected and logged in to this avaya, and i have a zoiper soft phone connected to asterisk |
| 23:47.23 | xochilpili | when i try to call from extension 1001 to extension 8000 using the prefix "7" in outbound route; i got this error in Cli: Unable to create channel of type 'OOH323' (cause 0 - Unknown) |
| 23:47.42 | xochilpili | please, can anyone give me a hand? |
| 23:59.27 | [TK]D-Fender | Show the full call attempt |