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| 07:08.12 | bittis | gdmorning guys |
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| 07:23.38 | drmessano | Diabittis |
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| 12:19.06 | Phil-Work | INVITE comes in to asterisk offering 3 codecs (18 0 8)... asterisk sends back an OK with 2 codecs (8 0) |
| 12:19.27 | Phil-Work | Asterisk starts sending PCMA (8), the caller starts sending PCMU (0) |
| 12:19.36 | Phil-Work | Asterisk realises and changes to PCMU and everything is ok |
| 12:19.50 | Phil-Work | but the start of the audio to the caller is lost, because it doesn't understand the codec |
| 12:20.24 | Phil-Work | this sounds ok, per the RFC, except perhaps that the caller should understand PCMA |
| 12:20.59 | Phil-Work | however, rather than arguing with a carrier, can Asterisk be made to only send a single codec in the response - that codec being the highest priority codec in the caller's INVITE |
| 12:21.07 | Phil-Work | that is, without disabling all bar one codec in Asterisk |
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| 12:38.51 | Phil-Work | nevermind, that was all totally irrelevant |
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| 12:44.02 | Chainsaw | You can generally transcode between PCMU & PCMA. It won't sound great, but it wouldn't cause a complete loss of audio. |
| 12:44.26 | Chainsaw | (But I take it you actually had some RTP establishment problem caused by NAT/firewalls?) |
| 12:45.39 | Phil-Work | it actually seems to be because Asterisk sends the RTP as soon as it's sent the 200 |
| 12:46.07 | Phil-Work | actually, that's not even true |
| 12:46.22 | Phil-Work | it sends it because it started receiving it |
| 12:46.48 | Phil-Work | though it's likely the 200 hadn't gotten through to the other end, as it's retramsitted... so the other side isn't entirely sure what codec it is |
| 12:47.06 | Chainsaw | You seem to only be looking at SIP here. What's the RTP story? |
| 12:47.22 | Phil-Work | there's no NAT |
| 12:48.15 | Phil-Work | the caller sends RTP about 0.7 seconds after it sends the INVITE in PCMU |
| 12:48.22 | Phil-Work | Asterisk then starts sending back in PCMU |
| 12:48.43 | Phil-Work | the caller appears to be dropping the first second or so of Audio - possibly because it hasn't gotten the 200 yet |
| 12:49.37 | Phil-Work | I'm thinking this is the caller's fault - if it is sending audio it should be able to receive audio back with the same codec even if it hasn't seen the 200 |
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| 14:15.17 | Samot | Retransmissions have nothing to do with the codec no being accepted. |
| 14:25.48 | Phil-Work | Samot, indeed |
| 14:26.07 | Phil-Work | but I suspect that the calling side isn't passing audio until the 200 has been received and processed |
| 14:26.29 | Phil-Work | it could well be for another reason - that just seems like the most likely given the timing |
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| 14:26.35 | Samot | Asterisk behind NAT? |
| 14:26.39 | Phil-Work | no |
| 14:26.52 | Phil-Work | I've reported it to the calling side to see what they think |
| 14:26.55 | Samot | Show a faile call. |
| 14:26.59 | Samot | failed |
| 14:27.16 | Phil-Work | can't share the pcaps, sorry |
| 14:27.51 | Samot | Then we can't help. |
| 14:28.07 | Phil-Work | that's fine - it's not an Asterisk issue |
| 14:28.36 | Samot | Are you sure? |
| 14:28.43 | Phil-Work | yes |
| 14:29.00 | Samot | So they just randomly ignore a reply to an INVITE to you? |
| 14:29.11 | Phil-Work | the audio is being sent to the carrier who originated the call - they are just not forwarding it on to the actual caller |
| 14:29.21 | Phil-Work | they don't ignore it - they just take a while to handle it |
| 14:29.25 | Samot | So the caller can't hear you but you can hear the caller? |
| 14:29.34 | Phil-Work | T1 is the standard 0.5 seconds - it's about 0.6 seconds before the 200 comes back |
| 14:29.56 | Phil-Work | hence Asterisk re-tramsits the 200 |
| 14:29.59 | Phil-Work | after that, the audio starts playing to the caller |
| 14:30.18 | Phil-Work | Asterisk is an IVR, in this instance |
| 14:30.33 | Phil-Work | it answers the call as soon as it gets the INVITE and starts playing some audio |
| 14:31.27 | Phil-Work | adding a short delay, or perhaps sending a 180 followed by a short delay, may well resolve it |
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| 15:08.04 | bittis | random question: is there any way of getting tasterisk to store logs in a database rather than files? |
| 15:09.12 | Samot | Not that I'm aware. |
| 15:09.26 | Samot | But the logs are formatted. |
| 15:09.30 | WIMPy | Use syslog on some PotOS. |
| 15:09.40 | Samot | I believe by tabs. |
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| 15:11.40 | bittis | cool, thanks guys |
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| 17:22.40 | tompaw | Is there a way to call a cross-server WebSocket endpoint? |
| 17:23.03 | tompaw | Or do I need an extra sip2ws context and do ServA->ServB->WS_endpoint? |
| 17:23.18 | tompaw | (I guess I need the latter, just making sure) |
| 17:29.49 | [TK]D-Fender | ? |
| 17:32.48 | tompaw | Right, I have a cluster of inbound servers which run ari bridges. They used to work with SIP Agent endpoints, which means that Server A could dial an agent registered on *ANY* other server by calling him directly (using SIP). |
| 17:33.13 | tompaw | Now we're migrating to websockets, as they are natively supported by the browsers and I don't have to fcuk about with zoiper applets/etc. |
| 17:33.41 | tompaw | However, that means that each Agent can only receive an inbound call *FROM* the server they are registered with (WS). |
| 17:34.18 | tompaw | So if a bridge on server A needs an agent who's on server B, server B must proxy this call (srvA---sip---srvB---ws---AGENT) |
| 17:34.21 | tompaw | Correct? |
| 17:34.59 | [TK]D-Fender | You never call a phone reg'd on another server directly... |
| 17:36.10 | tompaw | I used to, when they were SIP endphones, because all these asterisks shared the same psql contacts database. |
| 17:38.02 | [TK]D-Fender | EEK |
| 17:40.28 | igcewieling1 | tompaw: how many tens of thousands of calls do your servers process per day? |
| 17:40.44 | tompaw | 100k out, 3-5k in |
| 17:40.51 | igcewieling1 | per day? |
| 17:40.54 | tompaw | yep |
| 17:41.40 | tompaw | why? |
| 17:41.46 | igcewieling1 | nifty. You must be one of those people who actually really truly need more than one server. |
| 17:42.11 | tompaw | Yeah, I might need to squeeze all inbounds on 1 until I find a solution tho :-( |
| 17:42.37 | Samot | How many WS calls are you going to have? |
| 17:42.42 | tompaw | igcewieling1: it's a nice little setup with PXE diskless asterisks using casper and self-provisioning scripts. |
| 17:43.04 | igcewieling1 | tompaw: cool. a "hosted" service with "virtual PBXs"? |
| 17:43.07 | tompaw | Samot: if I migrate all my agents to WS, they would all be WS |
| 17:43.19 | Samot | That doesn't answer how many calls. |
| 17:43.20 | tompaw | igcewieling1: nope, bare metal |
| 17:43.29 | Samot | Are you saying that all 3-5K in will be WS? |
| 17:43.38 | tompaw | Samot: yes |
| 17:44.19 | tompaw | it's usually 10-50 inbound calls going on at the same time |
| 17:45.26 | Samot | Yeah, this is why I use NAPTR/SRV records. |
| 17:45.58 | tompaw | I use opensips in front of it all |
| 17:46.05 | Samot | I use Kamailio. |
| 17:46.16 | Samot | But I don't load balance my SIP registrations. |
| 17:46.37 | Samot | I also don't cram everyone on a single server. |
| 17:46.52 | tompaw | OK, so I guess what I need is for my ARI to distinguish if the agent it needs in a given bridge is from a local server or a remote one. If it's a remote -> sip2ws context via that agent's server. |
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| 19:04.42 | tompaw | Done and dusted. |
| 19:05.12 | tompaw | I wonder how much CPU is consumed by such a simple no-transcoding sip-to-ws call. |
| 19:05.16 | chris349 | Is there any way to create a hint/BLF to see the status of a phone registered to a different Asterisk server? |
| 19:08.49 | [TK]D-Fender | Yes. Google "asterisk distributed presence" |
| 19:11.21 | chris349 | Is there any way to use the existing SIP trunk instead of installing new applications? |
| 19:11.49 | chris349 | For e.g. theres already a way to subscribe via SIP to a remote MWI |
| 19:13.57 | [TK]D-Fender | no |
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| 19:31.56 | eric_hill | Does anyone know what version of asterisk introduced the cURL realtime back-end? |
| 19:36.40 | sekil | chris349: there is..use a proxy :) |
| 20:05.37 | sekil | btw.. |
| 20:05.56 | sekil | is asterisk able to work with kamailio/opensips as presence servers now? |
| 20:06.05 | sekil | I read something about this.. |
| 20:15.19 | *** join/#asterisk Phrohdoh (4222c1a6@gateway/web/cgi-irc/kiwiirc.com/ip.66.34.193.166) |
| 20:15.35 | Phrohdoh | What is invalid about the s exten? > [Dec 20 14:10:13] WARNING[43751][C-00000002]: pbx.c:4415 __ast_pbx_run: Channel 'SIP/kazoo-0000000d' sent to invalid extension but no invalid handler: context,exten,priority=kazoo-test-taryn,s,1 |
| 20:19.10 | Phrohdoh | Anyone? |
| 20:19.55 | [TK]D-Fender | "invalid extension" |
| 20:20.03 | [TK]D-Fender | does*not*exist |
| 20:20.59 | Phrohdoh | How does that not exist? |
| 20:21.16 | [TK]D-Fender | it's your dialplan |
| 20:21.22 | [TK]D-Fender | You told it tto go somewhere that you didn't make |
| 20:22.19 | [TK]D-Fender | If I tetll you to go to room 300, look for cabinet #4, and then opn the 4th drawer tto tto find something... and that cabinet doesn't exist... when then I sent you somewhere invalid |
| 20:22.52 | [TK]D-Fender | Same if you find cabinet #4 but it only has 3 drawers |
| 20:23.12 | [TK]D-Fender | That channel got sent somewhere that you didn't create |
| 20:23.34 | Phrohdoh | The condescension is a bit unnecessary, but ok. That dial plan does exist. I'll restart the service. |
| 20:23.56 | [TK]D-Fender | unless you just createtd i now restarting won't change anything |
| 20:24.04 | [TK]D-Fender | and you don'tt need to restartt the service to apply dialplan changes |
| 20:24.20 | [TK]D-Fender | you can do that live without disrupting calls via "dialplan reload" |
| 20:24.37 | Phrohdoh | Ah right, thanks. |
| 20:25.48 | Phrohdoh | Ok so the issue is that I had created it without applying dialplan changes, that makes sense. Now I'm being told my application doesn't exist so I'm willing to bet I just made a syntax error so I'll read about AGI again. |
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| 20:35.20 | Phrohdoh | So http://www.voip-info.org/wiki/view/Asterisk+AGI just says `script.agi`. My application is written in C# (my box has mono installed on it). How should i invoke my executable? |
| 20:35.33 | Phrohdoh | Everything I read about agi uses `AGI(something.agi,params)` |
| 20:36.35 | Phrohdoh | Ah my .agi should probably start with a shebang then I can do whatever. |
| 20:37.02 | roswell | Phrohdoh, just make it AGI(yourScript.net,params); # not sure whether mono allows shebangs |
| 20:37.30 | Phrohdoh | How do I tell asterisk to launch mono with my assembly's name as a param? |
| 20:37.44 | Phrohdoh | `mono myscript.exe "$@"` |
| 20:38.11 | roswell | Phrohdoh, make a wrapping shell script |
| 20:38.12 | [TK]D-Fender | You call an executable there |
| 20:38.38 | Phrohdoh | Right so my shell script would be a .agi with a (bash) shebang that I invoke mono in, ok |
| 20:39.04 | roswell | .agi is just a part of a filename |
| 20:39.52 | roswell | you can name it whatever you want. just make sure asterisk has permissions to launch it |
| 20:44.33 | Phrohdoh | Is it possible to just execute a dialplan instead of it executing when an incoming call matches the exten pattern? |
| 20:44.38 | *** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net) |
| 20:44.49 | Phrohdoh | Basically I'd like to be able to tell asterisk to run my dialplan (which will run my agi script) so my script can handle the rest. |
| 20:57.17 | [TK]D-Fender | Itt's you job to tell it what dialplan to execute |
| 20:57.24 | [TK]D-Fender | You ttell it where to go. |
| 20:57.40 | Phrohdoh | That's not what I'm asking. |
| 20:57.58 | Phrohdoh | I want to execute my dialplan on command not when an incoming call occurs. |
| 20:58.13 | [TK]D-Fender | You need a channel to execute dialplan. |
| 20:58.42 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
| 20:58.42 | [TK]D-Fender | Doesn'tt matter what kind |
| 20:59.10 | [TK]D-Fender | if you want a "non-person" type one just use a LOCAL channel |
| 20:59.40 | [TK]D-Fender | You'll have dialplan acting as the other side of the call, which could just sitting in a Wait(), etc while the other end executes relevent stuff, etc |
| 21:06.03 | *** part/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
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| 21:13.41 | igcewieling1 | Phrohdoh: if you need to execute schedule dialplan processing you'll need to create a call. You cannot do what you are trying to do without generating a call. Google for Asterisk's spool aka call files. |
| 21:13.55 | igcewieling1 | to schedule for the future you set the ctime of the resulting file to the future. |
| 21:14.00 | Phrohdoh | Yeah I've been spooling call files but was hoping to not do that. |
| 21:14.26 | igcewieling1 | you could write a daemon to connect to Asterisk using ARI then create the call. |
| 21:14.26 | Phrohdoh | The library I'm using to prototype (pycall) doesn't have docs for Application and Data (which I could use to run my dialplan). |
| 21:40.49 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
| 21:57.15 | DivideBy0 | igcewieling1's idea would be a 1-liner |
| 21:58.05 | Phrohdoh | I like the sound of that. |
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| 22:02.35 | DivideBy0 | not sure if it'd be much different from a call file if it's a 1-liner though |
| 22:02.49 | DivideBy0 | but you could add a few more lines and handle errors pretty easily |
| 22:08.33 | wyoung | hey NaN! |
| 22:17.47 | igcewieling1 | NaN? I hate that guy. Always messing up math. |
| 22:17.52 | igcewieling1 | 8-| |
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| 22:19.52 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:21.48 | DivideBy0 | hiya wyoung |
| 22:31.20 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
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| 22:52.07 | wyoung | igcewieling1: :P |
| 22:52.09 | wyoung | DivideBy0: :) |
| 22:52.32 | wyoung | What's crackin'? |
| 22:53.09 | *** join/#asterisk cervajs2 (~cervajs2@178.148.broadband4.iol.cz) |
| 22:53.33 | cervajs2 | hi, any news about astricon 2016 videos? |
| 22:54.10 | malcolmd | good question for Rusty Newton and/or David Duffett |
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