| 00:04.14 | *** join/#asterisk rpifan_ (~rpifan@ipb218f187.dynamic.kabel-deutschland.de) |
| 00:07.32 | *** join/#asterisk elcontrastador (~textual@70-90-215-98-BusName-ca.sacra.hfc.comcastbusiness.net) |
| 00:07.34 | Slade | Samot, neat i didnt know flowroute was actually a CLEC |
| 00:28.37 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-128-59.satx.res.rr.com) |
| 01:28.37 | *** join/#asterisk life_of_e (~life_of_e@108-95-189-245.lightspeed.irvnca.sbcglobal.net) |
| 01:38.25 | *** join/#asterisk rpifan (~rpifan@ipb218f187.dynamic.kabel-deutschland.de) |
| 02:02.37 | *** join/#asterisk mutin-sa (~s-mutin@85.234.114.134) |
| 03:13.26 | *** join/#asterisk bbt (~dbag@unaffiliated/bbt) |
| 03:29.14 | *** join/#asterisk bbt (~dbag@unaffiliated/bbt) |
| 03:46.12 | *** join/#asterisk yteltom (~textual@159-118-169-116.cpe.cableone.net) |
| 03:50.01 | *** join/#asterisk krzee (~k@openvpn/community/support/krzee) |
| 04:19.48 | *** join/#asterisk spresser (~astalsi@c-76-19-6-127.hsd1.ma.comcast.net) |
| 04:21.14 | spresser | Hey folks - I'm having an issue with the Console channel -- I can't get audio out of it, even though other channels (like SIP) work ok. Other applications play audio ok. What can I look at to determine where the audio is going/why I'm not hearing output? |
| 04:22.06 | spresser | (running 13.14.1 on Debian) |
| 04:41.41 | rockman37 | spresser: What is the value of output_device in console.conf? |
| 04:41.55 | rockman37 | spresser: And what audio system are you using? E.g., ALSA. |
| 04:42.34 | spresser | "default" And ALSA, I believe. Whatever the Debian default is. |
| 04:43.18 | rockman37 | Can you play audio with mplayer -ao alsa:device=hw=default or similar? |
| 04:43.27 | spresser | I certainly have the aplay, arecord, alsactl, etc commands and they work. |
| 04:44.52 | spresser | Yes |
| 04:45.11 | rockman37 | Goodoh. Are you loading chan_alsa.so in modules.conf? |
| 04:46.01 | spresser | Um, let me check. |
| 04:46.34 | spresser | Huh, I've got "noload => alsa" in there. |
| 04:46.51 | rockman37 | Try with load => and restart Asterisk. |
| 04:46.53 | spresser | Let me swap that for the oss noload and see what happens... |
| 04:47.03 | rockman37 | Cool. |
| 04:48.26 | spresser | Now I'm getting "No such extension 's' in context 'local' |
| 04:48.44 | rockman37 | What were you getting before? |
| 04:49.08 | spresser | No response, but if I tried "console dial" it theoretically created a call. |
| 04:49.23 | [TK]D-Fender | you are doing 2 very different things |
| 04:49.51 | rockman37 | [TK]D-Fender: Possibly, would you care to elaborate? |
| 04:50.23 | [TK]D-Fender | spresser, show what you actually just attempted that generated that message |
| 04:50.46 | spresser | [TK]D-Fender: demobox*CLI> console dial |
| 04:50.46 | spresser | No such extension 's' in context 'local' |
| 04:50.52 | [TK]D-Fender | You aren't dialing anything |
| 04:51.01 | [TK]D-Fender | you shouldn't just be leaving it incomplete like that |
| 04:51.55 | spresser | Sure. Trying it with an extension: |
| 04:51.55 | spresser | demobox*CLI> console dial 1000 |
| 04:51.55 | spresser | demobox*CLI> console dial |
| 04:51.55 | spresser | You're already in a call. You can use this only to dial digits until you hangup |
| 04:51.58 | spresser | Command 'console dial' failed. |
| 04:52.03 | spresser | But no audio. |
| 04:52.28 | [TK]D-Fender | what do you think should have happened? |
| 04:53.01 | spresser | My dialplan has a Milliwatt() that I should be hearing. |
| 04:53.10 | spresser | Also, it jsut spat out a couple hundred of this: |
| 04:53.11 | spresser | [Feb 13 23:49:48] ERROR[3027][C-00000000]: chan_alsa.c:486 alsa_read: Read error: Resource temporarily unavailable |
| 04:54.40 | [TK]D-Fender | is there something else that may have seized ALSA? |
| 04:54.57 | [TK]D-Fender | second part is the assumption of exactly what dialplan you assume it will hit |
| 04:55.31 | rockman37 | That error does sound like something else seized ALSA indeed. |
| 04:56.17 | [TK]D-Fender | "core set verbose 10" <- make sure we're turned up |
| 04:56.43 | spresser | Give me a minute and I'll reboot the box so nothing else grabs alsa. It brings up gdm, not the console. Should I have it go to text mode? |
| 04:57.05 | [TK]D-Fender | Are you running X on top of this? |
| 04:57.22 | spresser | Yep. Mostly by default. |
| 04:57.40 | [TK]D-Fender | Then I would avoid trying to use the console at all |
| 04:57.50 | [TK]D-Fender | install another SIP client and just use that |
| 04:58.01 | [TK]D-Fender | make sure it binds to a different port than * uses |
| 04:58.12 | [TK]D-Fender | so not 5060. perhaps 5065 |
| 04:58.24 | [TK]D-Fender | take your pick on which one. linphone, ekiga, etc |
| 04:58.30 | spresser | SIP works fine, but I need an audio output from this box's speakers. |
| 04:58.43 | spresser | Sounds like if I want to try that, I need to disable X? |
| 04:58.49 | [TK]D-Fender | yes, and using a GUI SIP client will use standard GUI sound resources |
| 05:00.02 | spresser | Alright, disabed X, rebooting |
| 05:01.08 | spresser | Hm, something is still grabbing ALSA apparently. |
| 05:01.36 | spresser | Same message. Also, the last message I get is " -- Executing [s@psst_demo:4] Milliwatt("ALSA/default", "") in new stack |
| 05:01.46 | spresser | So I think my dialplan is working as expected. |
| 05:02.07 | [TK]D-Fender | it's calling it... |
| 05:02.16 | [TK]D-Fender | so it's sorta getting somewhere. |
| 05:02.41 | [TK]D-Fender | console dial is one of the most dependent and PITA ways to try to test things |
| 05:02.51 | [TK]D-Fender | which is why I recommend just using your GUI and a SIP client on it |
| 05:03.04 | [TK]D-Fender | since it will do al the heavy lifting for you |
| 05:03.48 | spresser | Is there a command line/scriptable SIP client I could use? |
| 05:04.07 | [TK]D-Fender | You already were running a GUI as I believe you just said. Why not use it? |
| 05:04.45 | spresser | The eventual end goal involves being headless. The GUI was on there because Debian dropped it by default and I was too lazy to change it. |
| 05:05.05 | [TK]D-Fender | And you're working phsycially on that machine? |
| 05:05.15 | rockman37 | You could use pjsua on the command line, or probably a SIP module for Python or whatever. |
| 05:05.39 | spresser | For the moment, yeah, I'm on the machine. I have a headset connected to the 3.5mm port. |
| 05:06.11 | [TK]D-Fender | then if that's what you intend to test on then continue using the GUI and just use a friendly gui-based client |
| 05:06.36 | [TK]D-Fender | because the alternatives are just more pain that don't relate to real future use |
| 05:06.38 | *** join/#asterisk Zipper_32 (~Zipper@69.172.155.142) |
| 05:06.50 | rockman37 | Were you using the console just for testing, not because you actually want to use the console in production? |
| 05:07.00 | rockman37 | If so, I agree that's more trouble than it's worth. |
| 05:07.20 | spresser | Sorry - I'm not communicating well. I have my dialplan set and tested using ekiga. It's the "shifting to using locally-attached speakers (as opposed to ekiga on a separate machine)" step I'm currently stuck at. |
| 05:07.44 | [TK]D-Fender | I'm trying to verify why you are actually doing this |
| 05:08.04 | spresser | I don't care if it's Console or a SIP client in production as long as I can interact with it programmatically and it works with the local 3.5mm port. |
| 05:08.29 | [TK]D-Fender | This is kinda going in circles |
| 05:08.50 | [TK]D-Fender | " I have my dialplan set and tested using ekiga" <- you seem to have accomplished something this way |
| 05:08.59 | [TK]D-Fender | Why are you looking at ALSA right now? |
| 05:09.02 | rockman37 | What do you actually want to achieve by having audio over the soundcard on you PBX? |
| 05:09.15 | rockman37 | s/you PBX/your PBX/ |
| 05:11.20 | spresser | Hold on a sec while I come up with a relatively succinct and accurate explanation. |
| 05:11.28 | rockman37 | holds |
| 05:12.18 | [TK]D-Fender | wonders how hard this should be to express... |
| 05:13.13 | [TK]D-Fender | I'm getting the feeling there is something quite beyond this that hasn't been explained and is somehow pertinent |
| 05:13.48 | rockman37 | Possibly. |
| 05:14.05 | spresser | Yeah. This is all a demo for some CALEA-related tech and I have to be careful what I'm saying about that. |
| 05:14.16 | spresser | That may be the missing bit. |
| 05:14.37 | rockman37 | I don't know what CALEA is, apart from the plant. |
| 05:14.53 | [TK]D-Fender | we don't need to full usage, jsut the timing at which you actually NEED to use the local hardware |
| 05:14.59 | rockman37 | Oh, it's the wiretapping law, right? |
| 05:15.06 | spresser | rockman37: Yep. |
| 05:15.23 | [TK]D-Fender | so explain when you think you need console dial for anything. |
| 05:15.37 | spresser | This box is a demo of that tech. The concept is that demo users pick up a phone and interact with a simulated IVR. |
| 05:15.40 | Slade | huh messagebird seems pretty mature.. dunno if its the way to go in the US tho.. maybe flowroute still better.. that lookup api seems nice tho |
| 05:16.07 | [TK]D-Fender | I'm not sure what "tech" you're referring to |
| 05:16.23 | rockman37 | Picking up a phone and interacting with an IVR makes sense. |
| 05:16.23 | [TK]D-Fender | Describe the real order of actions. |
| 05:16.48 | rockman37 | Although I'm not sure what a "simulated" IVR would be. |
| 05:16.49 | [TK]D-Fender | X starts a process. Y happen. I need Z to happen next |
| 05:17.11 | [TK]D-Fender | This should be simple to explain. |
| 05:17.21 | *** join/#asterisk x5eb (~seb@seb-hpws2.w1.tele.crt1.net) |
| 05:18.43 | spresser | Ok. I have a phone connected to the 3,5mm jack. The use rpicks up the phone and asterisk starts running through the dialplan (I expect to write software to figure out when the phone is picked up, I know that's not asterisk functionality). Somehow audio from the phone gets through to asterisk and audio from asterisk gets played from the phone. |
| 05:19.00 | [TK]D-Fender | hold that though |
| 05:19.03 | [TK]D-Fender | thought |
| 05:19.14 | [TK]D-Fender | a phone doesn't connect through 3.5 and there is no "pickup" for * to detect |
| 05:19.18 | rockman37 | How does one connect a phone to a 3.5mm jack? |
| 05:19.42 | [TK]D-Fender | we need to rewind to the point at which a process doesn't actually "just work" |
| 05:19.43 | spresser | Very janky circuitry and a 600-ohm transformer. |
| 05:19.56 | rockman37 | That sounds bad. |
| 05:20.07 | [TK]D-Fender | Ok, you can run a custom hybrid for that. * still doesn't know about "pickup |
| 05:20.07 | rockman37 | Why not use some hardware designed for this, such as an ATA? |
| 05:20.13 | [TK]D-Fender | How do we get there? |
| 05:20.58 | Slade | messagebird seems much cheaper per minute than flowroute.. that might be suspicious |
| 05:21.05 | [TK]D-Fender | we need a trigger. |
| 05:21.05 | spresser | I'll write the software to detect the pickup. My plan was to have it call "asterisk -r 'console dial 1000'" when the phone was picked up. If the console worked, that would start the IVR doing what I want. |
| 05:21.54 | spresser | Re: why no ata: I could, but I need the 3.5mm anyway (it simulates a wiretap) |
| 05:22.16 | [TK]D-Fender | a plug type doesn't sould like the only definition or means. |
| 05:22.24 | [TK]D-Fender | depends what "wire" you're "tapping" |
| 05:22.39 | [TK]D-Fender | 3.5 is the interface you're forced to work with? |
| 05:22.53 | spresser | Yeah. Lowest common denominator. |
| 05:23.18 | [TK]D-Fender | is it jsut beacuse you think it should be used, or is it indeed forced on you? |
| 05:23.37 | [TK]D-Fender | "I've got this gear and that's the jack it's got.. so I have to |
| 05:24.05 | rockman37 | I wouldn't really want to buy a wiretapping device if it gave me only a 3.5mm jack. |
| 05:24.56 | spresser | It's the interface I've got that does raw audio. Other interfaces to be added later, the software works with raw audio _only_ (other protocols to be added later) |
| 05:25.25 | [TK]D-Fender | so on CLI-only level you'll also have to deal with hangup detection, etc. |
| 05:25.38 | spresser | Oh, I agree. Only a 3.5mm interface would be terrible for an actual product. |
| 05:25.47 | [TK]D-Fender | So I guess you'll have to sort out device contention for it |
| 05:26.47 | rockman37 | If it's Asterisk sending this audio to the telephone, why does it matter what the other software works with? |
| 05:27.32 | spresser | rockman37: The other software gets a copy of the audio of the call and does processing on it as the point of the demo. |
| 05:27.44 | [TK]D-Fender | This is going to be a very kludgy solution, but maybe you can make it work. |
| 05:27.55 | [TK]D-Fender | so get whatever is stealing access to sound out of the way |
| 05:28.09 | rockman37 | spresser: Ah, right. So you're essentially wiretapping a raw audio connection on the computer, rather than wiretapping any kind of phone line? |
| 05:28.49 | spresser | rockman37: Yes, for the moment |
| 05:30.01 | spresser | Maybe I should be asking a different question: How can I get 8-bit signed PCM audio at 8KHz out of Asterisk, including DTMF tones equivalent to any dialing done by the user? I can pick up and connect a SIP phone no problem as long as I can get that audio. |
| 05:30.15 | rockman37 | spresser: Hmm. So it's not (yet) useful for actually wiretapping a software PBX, given that they don't usually deal in raw audio. |
| 05:30.19 | rockman37 | But that's a side note. |
| 05:31.45 | rockman37 | You could probably use a call recording function, but you'd probably need it in realtime, right? I haven't played with recording calls in realtime. |
| 05:31.48 | [TK]D-Fender | audio out is really just audio. Are you referring to an encoded data stream holding the media of what would be audio? |
| 05:32.29 | spresser | rockman37: Plan is to send the audio to JACK or via EAGI. Haven't done my research on those bits yet. |
| 05:32.56 | spresser | [TK]D-Fender: I could work with that. As long as I can convert it to that format, I'm happy. |
| 05:33.11 | rockman37 | It sounds like we're coming up with a workaround to avoid using whatever software you're supposed to be demoing. |
| 05:33.30 | [TK]D-Fender | you still need to get audio in which was your goal here.. so lets just get that since you seemed to have the rest of the plan in place |
| 05:34.03 | [TK]D-Fender | Now that we understand the necessity and direction and hackery you know to expect let's try to get there |
| 05:34.15 | spresser | [TK]D-Fender: Sounds like a plan. Thank you. |
| 05:34.35 | rockman37 | I'd personally use a SIP phone or ATA and then get Asterisk to route the audio to wherever you need it once it's got into Asterisk. |
| 05:34.49 | rockman37 | That would save you the trouble of pickup detection and all that. |
| 05:35.06 | [TK]D-Fender | rockman37, The trick is the audio-in is coming from an odd place. That means detection, triggering, etc |
| 05:35.10 | [TK]D-Fender | we are WAY outside of the box. |
| 05:35.23 | rockman37 | [TK]D-Fender: But does the audio in have to come from an odd place? |
| 05:35.24 | [TK]D-Fender | "it is what it is" |
| 05:35.30 | spresser | rockman37: I'm leaning that way now, yeah. My big concern is the DTMF digits be in whatever output I get. |
| 05:35.41 | [TK]D-Fender | That's what he has confirmed so I'm taking that for what it is |
| 05:36.22 | spresser | I don't think the audio has to come from an odd place, that's merely the hardware setup I have (from an earlier stage). There's budget for some more hardware ($100s, not $1000s) |
| 05:36.25 | [TK]D-Fender | there is a potentioal issue here. I'm not sure if chan_ALSA does inband dtmf detection or not. |
| 05:36.36 | [TK]D-Fender | I've never had to think on this aspect |
| 05:36.46 | [TK]D-Fender | but lets see if we can get it to seize proerly at all first |
| 05:37.27 | rockman37 | spresser: What is the other end of your call (the one that isn't the telephone that someone picks up) going to be? |
| 05:38.19 | spresser | rockman37: The Asterisk dialplan, essentially. It prompts them for some info, reads it back, hangs up. The tech to be demo'd is essentially this weird third leg that doesn't interact with the other two. |
| 05:39.03 | rockman37 | I'd be tempted to conference everything together. |
| 05:39.21 | spresser | [TK]D-Fender: Looks liek chan_ALSA has no DTMF detection: https://community.asterisk.org/t/alsa-channel-and-dtmf-decode/71149/6 |
| 05:40.32 | rockman37 | I've just tested and it's possible to conference the console and two SIP phones together. |
| 05:40.55 | spresser | Maybe focusing on the live duplication of audio in the expected format aspect would be better? And I can just pick up an ATA or hardware SIP phone to avoid the whole issue with my weird fake hardware wiretap. |
| 05:41.01 | rockman37 | So by conferencing the console into the call, it seems like you should get your audio in the format your other program need it in, right? |
| 05:42.12 | spresser | rockman37: Wouldn't I want to conference in a chan_JACK for the audio? Seems like the console would just output to ALSA, which isn't really where I need the audio. |
| 05:42.44 | spresser | (or EAGI, instead of JACK) |
| 05:42.48 | rockman37 | spresser: If you can do that, sure. |
| 05:43.21 | spresser | I don't need the audio to go via the 3.5mm jack if it's being duped in Asterisk. The demo software can run on the same box. |
| 05:43.28 | spresser | (as the PBX) |
| 05:44.00 | rockman37 | What does the demo software actually do then? I previously thought its purpose was to snoop on the audio going to/from a soundcard. |
| 05:44.51 | spresser | rockman37: Not a sound card. It records information about interactions with IVR systems (at this point based on the raw audio) |
| 05:45.37 | rockman37 | spresser: Okay, so you can still demo it without cheating by just handing the audio to it? |
| 05:45.37 | spresser | The sound card was the easiest way to do a proof of concept, but SIP would work just as well, as long as I can get audio with DTMF in it. |
| 05:45.44 | spresser | rockman37: Yes. |
| 05:45.58 | rockman37 | Cool, no we're making progress :) |
| 05:46.17 | rockman37 | I think SIP will be a lot easier, given that SIP is designed for phones. |
| 05:46.41 | spresser | I think this turns the problem into "how do I get a copy of the audio of a call delivered to an external program in real time"? |
| 05:46.59 | spresser | rockman37: I'm getting that impression :) SIP seems like the way to go. |
| 05:47.17 | rockman37 | That's a pretty different and probably much more sensible problem to have. |
| 05:48.11 | rockman37 | Either play with JACK (I'm not experienced with JACK, but if Asterisk can talk to it, it should do what you want) or use Monitor() to generate some audio data that you can hand to the program. |
| 05:48.38 | spresser | Will I have to do any work to get DTMF equivalent to user dialing injected into the audio? |
| 05:49.05 | rockman37 | I'm not sure but I'll check since I'm curious. |
| 05:49.43 | spresser | Thank you! |
| 05:54.14 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 06:08.30 | rockman37 | spresser: It seems like you'll have to regenerate the DTMF tones to have them sent out to your other application. |
| 06:24.10 | [TK]D-Fender | If you want to process DTMF IN from that channel then I'd recomment using a CLI SIP client that you can set to "inband" |
| 06:24.46 | spresser | rockman37: Ok. I'll do some googling on how to do that tomorrow. |
| 06:25.05 | spresser | [TK]D-Fender: Thank you, I'll look for one I can do that with. |
| 06:31.25 | *** join/#asterisk perseiver (~gopalanay@122.160.98.244) |
| 07:21.05 | *** join/#asterisk pchero_work (~pchero@87.213.240.121) |
| 07:39.45 | *** join/#asterisk pchero_work (~pchero@51.247.195.35.bc.googleusercontent.com) |
| 08:05.57 | *** join/#asterisk Mr_Pleb_Mgoo (~jakeb@103.46.212.49) |
| 08:50.06 | *** join/#asterisk vandyk (~vandyk@189.63.147.146) |
| 09:05.32 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 09:13.03 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 09:24.17 | *** join/#asterisk rockman37 (~rockman37@122-60-43-242-adsl.sparkbb.co.nz) |
| 10:13.09 | *** join/#asterisk Downlots (~Downlots@185.73.41.1) |
| 10:14.15 | *** join/#asterisk Syco (~syco@86.43.91.148) |
| 10:20.40 | Syco | hi, a non technical question.. any news on astricon 2019? my company was looking at organizing and expedition, but there's still nothing on the website |
| 10:24.56 | rockman37 | Syco: Thats not until October, isn't it? |
| 10:27.10 | Syco | yes, but we need to plan flights from europe, going forward might be expensive |
| 10:27.39 | rockman37 | Afraid I can't help you there. |
| 10:32.17 | *** join/#asterisk Downlots (~Downlots@185.73.41.1) |
| 10:41.34 | *** join/#asterisk guerby (~guerby@april/board/guerby) |
| 10:44.48 | Syco | I know, I hoped someone on the channel had some inside news on the days |
| 10:44.51 | Syco | i'll just have to wait |
| 10:47.43 | *** join/#asterisk Downlots (~Downlots@185.73.41.1) |
| 10:54.06 | *** join/#asterisk pchero_work (~pchero@51.247.195.35.bc.googleusercontent.com) |
| 11:06.48 | *** join/#asterisk [J]oules (uid223833@gateway/web/irccloud.com/x-dpgaaeizhspmhdqr) |
| 11:13.43 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 11:13.50 | wyoung | o/ |
| 11:27.25 | rockman37 | \o |
| 11:28.21 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 11:35.08 | wyoung | sup rockman37 ? |
| 11:47.55 | *** join/#asterisk qxork (~qxork@unaffiliated/qxork) |
| 11:52.42 | *** join/#asterisk rpifan (~rpifan@ipb218f1d6.dynamic.kabel-deutschland.de) |
| 11:53.37 | *** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com) |
| 11:55.11 | rockman37 | wyoung: Not a lot, procrastinating this evening. |
| 11:55.14 | rockman37 | You? |
| 12:01.37 | Syco | On a different note.. I might need some help.. |
| 12:02.41 | Syco | I've a centos7, kernel 3.10.0-957.1.3.el7.x86_64, dahdi-linux-complete-2.11.1+2.11.1.tar.gz, libpri-1.6.0.tar.gz, asterisk-13.17.0.tar.gz, wanpipe-7.0.25.1.tgz |
| 12:02.58 | Syco | and I get this error: https://issues.asterisk.org/jira/browse/ASTERISK-16148 |
| 12:03.58 | Syco | I've never seen that in my whole 8 years working on asterisk and I can't find any solution/reason for it, do you know what could it be? |
| 12:31.56 | *** join/#asterisk MrMojit0 (~MrMojit0@87.213.99.78) |
| 12:42.10 | *** join/#asterisk rpifan (~rpifan@178.24.240.30) |
| 13:34.36 | *** join/#asterisk Ai9zO5AP (~BQcdf9eiZ@192.36.31.215) |
| 14:10.26 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:11.50 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 14:18.33 | *** join/#asterisk muAdmDev (~mu@193.158.65.130) |
| 14:22.37 | *** join/#asterisk billxx (47e9e6f3@gateway/web/freenode/ip.71.233.230.243) |
| 14:26.44 | *** join/#asterisk MLC (~MLC@63.249.40.11) |
| 14:41.59 | *** join/#asterisk pchero_work (~pchero@87.213.240.121) |
| 14:50.48 | *** join/#asterisk mahafyi (~mahafyi@182.65.85.39) |
| 15:02.46 | *** join/#asterisk kharwell (kharwell@nat/digium/x-dowjgwgljxqglhxy) |
| 15:02.46 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:15.20 | *** join/#asterisk pchero_work (~pchero@87.213.240.121) |
| 15:37.19 | *** join/#asterisk rpifan (~rpifan@ipb218f0d8.dynamic.kabel-deutschland.de) |
| 15:46.37 | *** join/#asterisk cresl1n (uid299068@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 15:46.37 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 15:50.54 | *** join/#asterisk josecapurro (~jcapurro@170.51.116.50) |
| 15:53.50 | *** join/#asterisk rpifan_ (~rpifan@ipb218f1ee.dynamic.kabel-deutschland.de) |
| 15:54.15 | josecapurro | Hello there! In version 11.25 and 1.8 i have this problem where when i try to call an SIP or IAX2 trunk, Asterisk says 34-Congestion and cannot do it. This goes away with a core restart now. |
| 15:54.24 | josecapurro | Any ideas? |
| 15:55.36 | josecapurro | All my boxes are full-IP. Only SIP and IAX2 trunks. |
| 16:01.58 | Samot | And neither one of those versions are supported anymore. |
| 16:04.58 | *** join/#asterisk jkroon (~jkroon@154.73.34.30) |
| 16:05.38 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-dmiblywswrsgimua) |
| 16:05.38 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 16:07.56 | *** join/#asterisk F29 (~F29@unaffiliated/bitcho) |
| 16:10.01 | *** join/#asterisk miralin (~Thunderbi@81.177.57.153) |
| 16:11.12 | *** join/#asterisk miralin1 (~Thunderbi@195.209.246.194) |
| 16:12.44 | josecapurro | kthx! |
| 16:15.18 | [TK]D-Fender | You'd have to actually show us the calls to compare |
| 16:15.35 | [TK]D-Fender | and "full ip" doesn't have any specific meaning here |
| 16:22.33 | Samot | Yeah, I got caught up in something else but my follow up was going to be "show what is happening" |
| 16:36.26 | *** join/#asterisk jkroon (~jkroon@154.73.34.30) |
| 16:51.01 | *** join/#asterisk aness (~aness@cm-84.209.49.44.getinternet.no) |
| 16:58.50 | *** part/#asterisk muAdmDev (~mu@193.158.65.130) |
| 17:08.46 | *** join/#asterisk miralin1 (~Thunderbi@81.177.57.153) |