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| 11:00.25 | [sID] | Hi, all |
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| 11:55.04 | [sr] | hi |
| 11:55.10 | [sr] | hi WIMPy |
| 11:57.11 | [sr] | damn, everybody's sleeping ! |
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| 12:07.52 | Arcticsilver | morning everyone, got an issue where voicemail just says "sorry" then diconnects? Anyone come accross this before? |
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| 12:14.30 | *** join/#asterisk hirogen1 (~hirogen@90.217.53.94) |
| 12:14.33 | hirogen1 | Hi |
| 12:14.55 | hirogen1 | any avaya experts ? |
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| 12:21.52 | Arcticsilver | know a little about avaya whats up? |
| 12:25.00 | hirogen1 | oh hey |
| 12:25.15 | hirogen1 | i suppose you havent come across avaya 1608 |
| 12:25.21 | hirogen1 | going completely dead |
| 12:25.30 | Arcticsilver | handset? |
| 12:25.32 | hirogen1 | PoE |
| 12:25.32 | hirogen1 | yeah |
| 12:25.46 | hirogen1 | sometimes tehy might work again after a few days, ive googled the life out of it |
| 12:26.26 | Arcticsilver | you tried it with a psu and not poe? |
| 12:26.38 | hirogen1 | dont worry too much I suppose as I want to learn how to deploy the one-x communicator software phone instead, via GPO though thats more windows related question i suppose |
| 12:27.00 | hirogen1 | Arct you know thats worth a shot |
| 12:27.01 | Arcticsilver | thats how we deploy it but we use sccm also |
| 12:27.09 | hirogen1 | we dont have many psu for them if any but sure its worth a try |
| 12:27.30 | Arcticsilver | should rule out any poe issues |
| 12:27.49 | hirogen1 | though to be fair this one guy on friday |
| 12:27.59 | hirogen1 | had a psu in his because he said it didnt work otherwise |
| 12:28.19 | hirogen1 | and I tried to remote connect to his computer to fix a provlem on his pc, and he said the phone went dead even though it was plugged in |
| 12:28.36 | hirogen1 | i was trying to remote connect to his pc using some remote tool lol |
| 12:28.44 | hirogen1 | its almost like the extra packets caused the phone to die |
| 12:28.47 | Arcticsilver | dead on psu or poe? |
| 12:28.48 | hirogen1 | who knows |
| 12:28.51 | hirogen1 | both |
| 12:29.15 | hirogen1 | its always conencted via poe as the lan cable powers the phone first then provives a network connection to the pc |
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| 12:29.26 | Arcticsilver | we have 1400 9608's and we doget a few the do that |
| 12:29.28 | hirogen1 | its apparently a known issue with these phones |
| 12:30.31 | Arcticsilver | i would say so! |
| 12:31.12 | Arcticsilver | im struggling with an asterisk box just saying sorry then disconnecting when going to vm :( |
| 12:32.09 | hirogen1 | also even when these phones are working the display has funny egyption characters and most of the time when the display is actually normal it doesnt quite show the full extension number. so its like the P0e power doesnt fully charge the phone i suppose |
| 12:33.05 | Arcticsilver | not on long ethernet are they? |
| 12:36.00 | hirogen1 | not sure |
| 12:36.06 | hirogen1 | not at work to check define long ? |
| 12:40.05 | hirogen1 | Im going to figure out how to install a software phone I got it working on my computer and a few others but I need to figure out how to deploy the x-one communicator via Group policy to all machines, thats a good work around then we can scrap the phones |
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| 12:42.47 | Arcticsilver | if your using one x client you might need to check your network config as you may not have QOS enabled on data vlan if you have a voice vlan configured, the switch will automatically drop the phone in the voice vlan etc |
| 12:42.55 | AL13N | i don't understand... i have zero tls anywhere in asterisk, it's all configured, but no ports are opened... (in netstat -lntup ) ... not in http.conf ; not in sip, nothing... how is this failing? |
| 12:43.27 | AL13N | and asterisk is compiled with -lssl |
| 12:43.46 | hirogen1 | articsilver I can confirm it works |
| 12:43.55 | hirogen1 | we've got 4 ppl in It using it |
| 12:43.58 | Arcticsilver | :) hiro |
| 12:44.03 | hirogen1 | plus 2 users |
| 12:44.04 | hirogen1 | hehe |
| 12:44.23 | hirogen1 | so that aspect must have been done by the avaya guy who came in last year before i started, im just a temp guy anyway in It support lol |
| 12:44.32 | hirogen1 | now they kinda hinted they want me to role it out |
| 12:44.38 | Arcticsilver | it will work, just might not recognise the dscp markings |
| 12:45.01 | AL13N | i was trying to get FF with webrtc and dtls to work... but now i have no extra port open... |
| 12:45.07 | hirogen1 | only bad thing was they dont have or lost the Avaya ID account so i cant d/l the latest avaya communicator lol |
| 12:45.11 | AL13N | i set tlsenable=yes and dtlsenable=yes |
| 12:45.18 | AL13N | put in certificates and everything |
| 12:45.30 | hirogen1 | i inputted our avaya server address and it picked up the h.232 and login extension works fine |
| 12:45.34 | hirogen1 | so the back end is all setup i suppose |
| 12:45.52 | hirogen1 | though x-one sp5 comes with extra stuff i need to lock it down |
| 12:46.01 | Arcticsilver | pm me i may have the latest version at work, what cm ver are you on? |
| 12:46.06 | hirogen1 | i dont want to intergrate it with browsers and outlook because we use lync |
| 12:46.16 | hirogen1 | thanks |
| 12:46.24 | hirogen1 | i think 6.2 |
| 12:46.41 | Arcticsilver | there is a version the integrated with lync, i got the new beat for that too |
| 12:46.41 | hirogen1 | we needed a newer version cos it didnt install right on 64 bit o/s though too be fair we do mostly use 32 bit o/s |
| 12:46.52 | hirogen1 | wow cool |
| 12:47.05 | hirogen1 | i suppose you aint got instructions on deploying it to all machines ? |
| 12:47.10 | AL13N | but, i also don't get any errors anywhere |
| 12:47.13 | AL13N | how can i debug this? |
| 12:47.21 | Arcticsilver | i do but on sccm not via gp |
| 12:47.32 | AL13N | tried starting with -cddddddddddvvvvvvvvvvv but nothing that points to this |
| 12:47.53 | hirogen1 | ah ok |
| 12:48.02 | hirogen1 | yeah our place wont go sccm for atleast another year |
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| 15:12.39 | xpheres | hi |
| 15:13.02 | xpheres | does anyone a way to make work webrtc with asterisk? |
| 15:14.46 | xpheres | does anyone knows a way to make work webrtc with asterisk? |
| 15:15.20 | xpheres | I'm able to register but there's no way to make a call |
| 15:18.02 | xpheres | if anyone has achieved this I would be very grateful for advice |
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| 15:30.31 | lnb | looking at this page, http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-4-SECT-7.html#asterisk-CHP-4-FIG-4 The doc says to set NO for nat mapping and nat keep alive when configuring a linksys spa942. The doc doesn't specify if the endpoint is on same subnet or remote. |
| 15:31.19 | lnb | since our pbx's are all in remote data centres, will the endpoint loose registration? |
| 15:32.26 | lnb | the reason i ask this is the amount of bandwidth used by what is generated by notify from a bunch of end points. I want to reduce this as much as possible |
| 15:44.27 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 15:47.08 | WIMPy | Hi [sr] |
| 15:47.39 | WIMPy | lnb: If you need keep-alive, use keep-aliv, don't abuse OPTIONS for that. |
| 15:50.56 | lnb | hi WIMPy |
| 15:51.12 | lnb | the truth of the matter is, I do not know if keep-alive is needed |
| 15:51.30 | lnb | the phones are in offices, the pbx is in remote data center |
| 15:51.49 | lnb | what i see in cli are non stop notify messages |
| 15:52.01 | lnb | and those use a ton of bandwidth |
| 15:52.32 | lnb | the phones send this message about every 3 to 4 seconds |
| 15:53.03 | WIMPy | Wow. That's a lot. |
| 15:59.36 | lnb | so the question remains |
| 15:59.56 | lnb | when does nat keep-alive have to be turned on? |
| 16:00.36 | WIMPy | When the router might forget about the "connection". |
| 16:00.39 | [TK]D-Fender | When you need a keep-alive and the OTHER SIDE is going to do a better job of it than your phone. |
| 16:00.53 | WIMPy | Which is round about always. |
| 16:01.17 | WIMPy | Depends on the router how fast they forget. |
| 16:03.59 | lnb | hmm |
| 16:04.08 | [TK]D-Fender | yup. I've never seen one that would require 3-4 seconds |
| 16:04.26 | [TK]D-Fender | So try testing the other way and not being stupid, and jsut see what the hell happens :) |
| 16:04.40 | WIMPy | I have had one where 20s only worked most of the time. |
| 16:04.46 | lnb | from the time the spa942 is powered on, how long would one wait to see if reg is lost? |
| 16:04.56 | [TK]D-Fender | lnb, not applicable <- |
| 16:05.01 | lnb | no? |
| 16:05.08 | [TK]D-Fender | this has NOTHING to do with registration |
| 16:05.08 | WIMPy | You wouldn't. |
| 16:05.30 | lnb | oh, this is strictly for nat |
| 16:05.44 | WIMPy | That's the evil thing about it. If you try to find out if it still works, you will reset the time (hopefully). |
| 16:06.18 | lnb | if i get this, if someone called the extension from outside (pots phone somewhere) and it has lost the nat connection, call would go to voice mail.. phone would not ring |
| 16:06.21 | WIMPy | So you qualify on asterisk an try to increase times at both ends. |
| 16:06.35 | WIMPy | yes |
| 16:07.41 | lnb | ok so if the setting for nat mapping and nat keep-alive was set to no, and a call came in 12 or 13 hours later. phone rings. then it doesnt need those set to yes |
| 16:07.50 | [TK]D-Fender | Not the "times" per-se. The frequency |
| 16:08.07 | lnb | i dont follow you.. |
| 16:08.14 | lnb | there is no frequency setting |
| 16:08.20 | WIMPy | The phone would register more than once every 12 hours. |
| 16:08.28 | lnb | if there was i would set it to 3600 |
| 16:08.30 | WIMPy | There should be. |
| 16:08.52 | lnb | i will take a screen shot and paste it |
| 16:09.09 | WIMPy | That's most probably too much. Something in the way of 60s might be realistic. |
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| 16:20.01 | lnb | http://www.servaris.com/images/voip/keep-alive.png |
| 16:20.23 | lnb | there is no setting that i see for frequency of keep-alive messages |
| 16:20.36 | lnb | its either on or off |
| 16:21.08 | WIMPy | That's bad. |
| 16:21.21 | lnb | thats sipura/linksys |
| 16:21.46 | lnb | 37 phones like this |
| 16:21.57 | lnb | it uses 320gb / month |
| 16:22.01 | WIMPy | I know. that design. |
| 16:22.09 | [TK]D-Fender | So don't use it |
| 16:22.16 | lnb | its not mine |
| 16:22.22 | lnb | its a clients phones |
| 16:22.26 | lnb | they like them |
| 16:22.38 | [TK]D-Fender | THE KEEP-ALIVE ON THE PHONE <---- |
| 16:22.38 | lnb | i am just trying to save a bad situation for us |
| 16:22.43 | lnb | oh |
| 16:22.46 | lnb | sorry |
| 16:22.47 | WIMPy | But you can change the message. Maybe to null or something. |
| 16:23.08 | [TK]D-Fender | "Doctor, doctor, it hurts when I raise my arm like this...." |
| 16:23.12 | lnb | i am going to watch the log for a minute and see exactly what code its sending |
| 16:23.40 | lnb | doctor to patient -> cut off arm. problem solved |
| 16:24.46 | WIMPy | You seem to have experiences with doctors. |
| 16:25.02 | lnb | millions of these: |
| 16:25.06 | lnb | Event: keep-alive |
| 16:25.07 | lnb | User-Agent: Linksys/SPA942-6.1.5(a) |
| 16:25.32 | lnb | WIMPy: just playing on [TK]D-Fender |
| 16:25.38 | lnb | statement |
| 16:25.51 | WIMPy | Sounded realistic, tho. |
| 16:26.10 | lnb | the question is, if those settings are set to no, will the phones loose connectivity? |
| 16:26.36 | WIMPy | Yes |
| 16:26.46 | lnb | great |
| 16:26.53 | WIMPy | You might get away with turning on qualify on asterisk. |
| 16:27.03 | lnb | qualify is on |
| 16:27.10 | WIMPy | If you set that to 60s or so that should be fine. |
| 16:27.41 | WIMPy | Usually that is the bad solution, but in your case it's better than what you have. |
| 16:27.55 | lnb | here is another big issue. Sometimes in this office, the phones line LEDs will go orange. For no apparent reason. |
| 16:28.05 | lnb | they took 7 of those phones off line |
| 16:28.11 | lnb | i have them here at my office |
| 16:28.22 | lnb | i have been testing them since thursday |
| 16:28.27 | WIMPy | What does orange mean? |
| 16:28.41 | lnb | it means you cant call or get calls |
| 16:29.01 | lnb | since thursday i cannot get the phones to go orange again |
| 16:29.15 | lnb | that tells me there is something wrong in the clients network |
| 16:29.15 | WIMPy | If the phone was able to find out about it, you might have a bigger issue. |
| 16:29.30 | lnb | huh? |
| 16:29.39 | WIMPy | Might be one of those evil routers that forget at random times. |
| 16:29.48 | lnb | ahhh |
| 16:29.52 | lnb | thats what i am thinking |
| 16:30.09 | lnb | the client has an 'IT' guy that loves crap hardware |
| 16:30.25 | lnb | i use a mikrotik router here |
| 16:30.27 | WIMPy | If that's the issue there isn't much yu can do about it. |
| 16:30.42 | lnb | and i cannot get any of the 7 phones to go orange |
| 16:30.46 | lnb | which is good |
| 16:31.07 | lnb | WIMPy: you hit that right on |
| 16:31.44 | lnb | so would it be safe to assume the crap router is loosing state? maybe on dhcp? |
| 16:31.53 | WIMPy | Well, not much = nothing. |
| 16:32.24 | WIMPy | Possible, but it's almost certainly the connection tracking. |
| 16:32.51 | lnb | connection tracking is part of what? |
| 16:33.03 | WIMPy | NAT |
| 16:33.10 | lnb | ahhh |
| 16:33.32 | lnb | so that might mean the router (pfsense) is doing a crap job with nat |
| 16:33.48 | WIMPy | Most probably. |
| 16:34.00 | lnb | thank you so much WIMPy! |
| 16:34.02 | lnb | finally |
| 16:34.08 | WIMPy | Maybe it just runs out of memory. |
| 16:34.24 | WIMPy | Is pfsense linux or bsd? |
| 16:34.33 | lnb | bsd |
| 16:34.51 | WIMPy | Don't know how memory allocation is done there. |
| 16:34.52 | lnb | you can put pfsense on anything |
| 16:35.15 | lnb | and in this case, the IT guy buys the cheapest CRAP that can be found |
| 16:35.36 | WIMPy | I guess that's what they got. |
| 16:35.43 | Penguin | The hardware doesn't really impact connection tracking. |
| 16:35.48 | lnb | well they point the finger at me |
| 16:35.58 | WIMPy | But they can still place outbound calls. Not too bad :-) |
| 16:36.10 | lnb | they used to have all these phones going to pbx -> telco |
| 16:36.25 | lnb | elastix pbx -> telco |
| 16:36.29 | WIMPy | With default configuration, the hardware does matter. |
| 16:36.36 | lnb | sure it does |
| 16:36.41 | [sr] | lnb: it isn't something with the same IP of the pbx ? |
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| 16:36.59 | lnb | sr? |
| 16:37.12 | Penguin | What is default configuration of pfSense? The hardware isn't related. |
| 16:37.20 | lnb | no clue |
| 16:37.23 | lnb | they wont show me |
| 16:37.47 | Penguin | Default configuration doesn't really work for much. It has to be configured to do what your network needs to do. |
| 16:37.54 | lnb | i suggested to the owner of the company to ditch the pfsense and get a mikrotik and the problems in the network would end. |
| 16:38.12 | WIMPy | Penguin: Not pfsense. But Linux default to only using a small fraction of available RAM for connection tracking, even if more is available. |
| 16:38.48 | Penguin | I don't think pf has that problem. |
| 16:38.48 | lnb | i use PF on a couple of FreeBSD Servers and it works well for firewall |
| 16:39.14 | lnb | but never tried to replicate what a mikrotik router can do with PF |
| 16:39.25 | Penguin | pf is good, but when you're using pfSense, you have to configure it well for work with VoIP. |
| 16:39.31 | WIMPy | It surely depends on the hardware and the number of connections it has to track. |
| 16:42.10 | Penguin | pf relies on hardware just like every other software relies on hardware. |
| 16:42.37 | lnb | i couldn't agree more |
| 16:42.44 | Penguin | It's not special. |
| 16:43.06 | lnb | its just like memory.. most people think memory is all the same |
| 16:43.11 | lnb | and that is incorrect |
| 16:43.41 | WIMPy | It can help to play with contrack timeout settings. |
| 16:43.41 | Penguin | But it isn't a case of "this motherboard isn't as good of quality as the one I could have bought, so pf won't work right." That isn't how things work. |
| 16:43.59 | lnb | we always install samsung memory for our server side business for clients |
| 16:44.12 | lnb | mother boards too are extremely important |
| 16:44.24 | lnb | for servers especially |
| 16:44.36 | Penguin | But pf doesn't work better or worse depending on which one you buy. |
| 16:44.40 | Penguin | It just works. |
| 16:44.52 | Penguin | If the board fails, everything software fails with it. |
| 16:45.11 | lnb | if you have faulty components, you will have strange problems creep up |
| 16:45.25 | Penguin | So in that sense, yes pf relies on hardware. But that isn't important in this discussion. |
| 16:45.42 | lnb | Penguin: what you are stressing is configuration |
| 16:45.49 | lnb | and that is also very true |
| 16:45.53 | Penguin | Look at the NAT configuration. |
| 16:46.03 | lnb | i wish i could |
| 16:46.09 | lnb | 'access denied' |
| 16:46.40 | Penguin | Every pfSense problem that has popped up in here with regard to asterisk has been related to the NAT configuration. |
| 16:47.24 | WIMPy | Setting up a firewall takes someone who understands the configuration. |
| 16:47.38 | Penguin | I don't know if pfSense is trying to be clever or the people running it have tried to be clever, but someone did something wrong time and time again. |
| 16:52.03 | lnb | in that regard its the same as setting up asterisk. if its not setup right, there will be all sorts of problems |
| 16:52.45 | [TK]D-Fender | #captainobvious |
| 16:52.51 | WIMPy | sterisk is like Windows 95. It works most of the time. If you don't touch it and didn't ask too much of it in the first place. |
| 16:56.35 | *** join/#asterisk Akuma (~Akuma@modemcable221.82-177-173.mc.videotron.ca) |
| 17:13.33 | lnb | under SIP tab on this spa942 phone... NAT Keep Alive Intvl: 15 |
| 17:13.57 | lnb | at least i found it |
| 17:14.07 | lnb | but its set to NO anyway |
| 17:14.13 | Penguin | It should be. |
| 17:14.27 | Penguin | In almost every case using asterisk, you disable all nat settings on the phone. |
| 17:14.56 | lnb | nat is needed if the pbx is remote isnt it? |
| 17:15.41 | [TK]D-Fender | This is not normally for the PBX |
| 17:15.43 | WIMPy | There's a lot of stuff related to NAT that can be configured at both ends. |
| 17:15.52 | [TK]D-Fender | This is to keep the PHONE side alive |
| 17:15.59 | [TK]D-Fender | Phones don't maintain PBX's |
| 17:16.04 | lnb | right |
| 17:16.05 | [TK]D-Fender | You're living in backwards-world |
| 17:16.13 | lnb | ok |
| 17:16.44 | lnb | in all cases every extension is set for quality every 60 seconds |
| 17:16.53 | lnb | s/quality/qualify |
| 17:17.01 | Penguin | Extensions don't have qualify, devices do. |
| 17:17.21 | lnb | in extension settings |
| 17:17.41 | WIMPy | If you're concerned about traffic, why do you enable qualify? |
| 17:18.10 | WIMPy | Are you talking FreePBX or something? That's not Asterisk talk. |
| 17:18.26 | lnb | oh ok |
| 17:18.28 | lnb | yes |
| 17:18.55 | lnb | qualify is set by default |
| 17:19.00 | lnb | qualify=yes |
| 17:19.00 | lnb | qualifyfreq=60 |
| 17:19.12 | [TK]D-Fender | Then doing both is plain stupid |
| 17:19.25 | lnb | meaning on endpoint ? |
| 17:19.32 | lnb | agreed if thats what you mean |
| 17:19.41 | lnb | but i cannot change it |
| 17:19.54 | lnb | unless i drive down to the office |
| 17:20.02 | lnb | but then that router will screw it up |
| 17:20.31 | lnb | its like WIMPy said, there is nothing i can do |
| 17:20.39 | [TK]D-Fender | ... |
| 17:20.40 | [TK]D-Fender | no |
| 17:20.42 | Penguin | What kind of weird world do you live in where you cannot remotely manage your asterisk? |
| 17:20.44 | lnb | but these 7 phones are going to a new office |
| 17:20.52 | [TK]D-Fender | those are BOTH keep-alives |
| 17:20.54 | lnb | Penguin: i manage the pbx |
| 17:20.55 | [TK]D-Fender | You ahven't been paying attention |
| 17:20.57 | [TK]D-Fender | pick ONE |
| 17:20.59 | [TK]D-Fender | TUNE it |
| 17:21.14 | lnb | Penguin: i can not vpn to the office network |
| 17:21.21 | lnb | where the endpoints are |
| 17:21.47 | Penguin | I guess ssh is broken as well. |
| 17:21.54 | lnb | Penguin: no |
| 17:22.00 | lnb | access denied |
| 17:22.09 | lnb | the IT guy will not allow me access |
| 17:22.49 | lnb | last week while in the office the owner asked me if the 7 phones sitting on a desk should be thrown out or can they be fixed. |
| 17:22.59 | Penguin | If you aren't the IT guy, I probably wouldn't either. |
| 17:23.12 | lnb | i said i would bring them to my office and see if i can get them to work |
| 17:23.33 | lnb | so i brought the 7 phones here |
| 17:23.47 | lnb | configured them to use the same PBX as the office does |
| 17:24.02 | lnb | they all work perfectly. No 'orange' LEDs |
| 17:24.13 | lnb | made lots of calls in/out |
| 17:24.29 | lnb | this video is disgusting |
| 17:25.51 | lnb | i just emailed the owner and told them. The problem is your routers. |
| 17:33.55 | lnb | [TK]D-Fender: for the record, i did not setup the settings on the SIP tab page of this particular phone. So dont assume that I set it up incorrectly. All i was asking originally was, if keep-alive for nat is needed. Which has be very well answered. |
| 17:35.03 | [TK]D-Fender | If you left qualify on for it than that is your fault |
| 17:35.16 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 17:35.24 | [TK]D-Fender | <lnb> in all cases every extension is set for quality every 60 seconds |
| 17:36.35 | Penguin | <lnb> qualify=yes <------- this equals 2000, not 60 |
| 17:37.12 | WIMPy | Wrong parameter. |
| 17:37.42 | Penguin | The default value of qualifyfreq is 60, and it is usually a good enough value. |
| 17:38.29 | WIMPy | It still shouldn't be abused as a replacement for client side keep-alive. |
| 17:39.58 | [TK]D-Fender | It should be used a substitute for a lack thereof... of for one that is abusive |
| 17:40.08 | [TK]D-Fender | Like on the order of <5 seconds. |
| 17:44.40 | lnb | qualify=yes is in the sip settings in pbx |
| 17:44.59 | lnb | on the endpoint i have changed nat keep-alive to no |
| 17:45.15 | [TK]D-Fender | If you are doing that and they have that keep-alive on you are wasting your own BW for nothing |
| 17:45.19 | [TK]D-Fender | that is a DOUBLE keep-alive |
| 17:45.48 | lnb | earlier today i said there was no frequency. when i looked in other tabs of the phones gui, the setting was found for keep-alive freq. |
| 17:46.34 | lnb | keep-alive=no in endpoint so how is that double? |
| 17:47.40 | [TK]D-Fender | <lnb> millions of these: |
| 17:47.40 | [TK]D-Fender | <lnb> Event: keep-alive |
| 17:47.40 | [TK]D-Fender | <lnb> User-Agent: Linksys/SPA942-6.1.5(a) |
| 17:47.54 | Penguin | (1244.59) <lnb> on the endpoint i have changed nat keep-alive to no |
| 17:47.59 | Penguin | changed to "no" |
| 17:48.49 | lnb | yes |
| 17:48.51 | lnb | correct |
| 17:49.11 | lnb | so if the endpoint has nat keep-alive=no whats wrong? |
| 17:49.34 | lnb | when i plugged in these 7 phones, every one was set to nat keep-alive=yes |
| 17:49.42 | lnb | that was the setting set by the IT guy |
| 17:49.46 | Penguin | That's fairly typical. |
| 17:50.18 | Penguin | Everyone seems to think the phone should do it... except for the people who actually use asterisk. |
| 17:50.39 | lnb | well the document on asteriskdocs.org that i found when searching should nat keep-alive be on or off, says no |
| 17:51.00 | lnb | and thats when i asked in here |
| 17:51.21 | lnb | but for some reason [TK]D-Fender thinks i am setting that value to yes |
| 17:51.36 | [TK]D-Fender | You had it at yes at the same time as qualify |
| 17:51.38 | [TK]D-Fender | Maybe not NOW |
| 17:51.45 | lnb | ok no problem |
| 17:51.50 | [TK]D-Fender | and I was typing my line as you were typing your second |
| 17:51.58 | lnb | no problem |
| 18:00.43 | *** join/#asterisk gusto (~gusto@2a02:810d:a840:3c20:82fa:5bff:fe0a:dfef) |
| 18:06.45 | WIMPy | Off course the phone should do it. But not the way it was done there. |
| 18:14.05 | WIMPy | You enable qualify if you need to know the state of the peer. |
| 18:34.59 | *** join/#asterisk demonspork (~demonspor@173-202-176-125.dyn.centurytel.net) |
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| 19:31.05 | *** part/#asterisk riess82 (~riessma@93-82-69-49.adsl.highway.telekom.at) |
| 19:37.27 | *** join/#asterisk sruffell (sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:37.27 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:41.49 | *** join/#asterisk bmurt (~brendan@64-121-3-32.c3-0.upd-ubr2.trpr-upd.pa.cable.rcn.com) |
| 19:48.48 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
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| 20:38.58 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
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| 20:42.45 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 21:06.14 | *** join/#asterisk skirmisha (~vk@46.47.82.189) |
| 21:06.24 | skirmisha | hi guys |
| 21:07.04 | skirmisha | any ideas why asterisk is not seeing some sip msgs? i see them on tcpdump, but they are not going to asterisk itself, not even single log |
| 21:07.17 | skirmisha | what could be the reason for that? |
| 21:07.37 | pjensen00 | What do you mean by "some sip messages"? |
| 21:07.47 | pjensen00 | you see invites coming in but not seeing followups? |
| 21:09.28 | skirmisha | i am trying to register a phone, i see other accounts are registered, server replies, but on my msg of my client nothing happens, no reply |
| 21:09.40 | [TK]D-Fender | Show us |
| 21:10.03 | skirmisha | when i dump on linux level i see my msg are coming/going to server, but not seen in asterisk log whatsoever |
| 21:10.29 | pjensen00 | You see absolutely no traffic in asterisk for that one endpoint? |
| 21:10.39 | skirmisha | restarted asterisk and still the same |
| 21:10.51 | skirmisha | yes no traffic for that extension |
| 21:11.28 | skirmisha | is there anything like module that could block this? |
| 21:11.53 | pjensen00 | When you say other accounts are registered, are they endpoints of the same type? |
| 21:12.24 | skirmisha | yes, same sip type of accounts |
| 21:13.03 | pjensen00 | I mean the endpoint you're trying to register outside of asterisk. Softphone/sip client. |
| 21:13.13 | pjensen00 | the physical one |
| 21:13.13 | skirmisha | i tried to create diff account names and no luck, this is first time i see such behavior |
| 21:13.29 | skirmisha | it is softphone |
| 21:13.59 | pjensen00 | Can you show the register that you're capturing with the tcpdump of the endpoint who is having the trouble? |
| 21:14.03 | skirmisha | it is like msg is not catch by ast appl |
| 21:14.14 | skirmisha | yes sure |
| 21:14.16 | pjensen00 | in pastebin or something? |
| 21:17.37 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 21:17.37 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:18.09 | WIMPy | Do you see those message in Asterisk if you enable sip debug? |
| 21:19.08 | skirmisha | no, debug is not showing anything |
| 21:19.32 | skirmisha | is there a code that asterisk ignore the msg if some fields are missing |
| 21:19.46 | skirmisha | i compare what my client sends what what the other sends |
| 21:57.15 | *** join/#asterisk theron (~theron@173-20-126-202.client.mchsi.com) |
| 22:20.14 | phix | Is there a man (or women) in the middle? |
| 22:21.56 | [TK]D-Fender | He's long gone... |
| 22:26.04 | phix | ah, he is one of those people |
| 22:26.26 | phix | "OH I have an issue, you have to help me! hurry up and help! <disconnects>" |
| 22:26.42 | phix | How you been btw [TK]D-Fender ? |
| 22:28.06 | [TK]D-Fender | "meh" |
| 22:28.42 | phix | uh oh |
| 22:29.22 | phix | Feeling a bit indifferent or is the prozac kicking in? |
| 22:32.17 | *** join/#asterisk ThatDamnRanga (~wiretap@unaffiliated/wiretap) |
| 22:35.48 | *** join/#asterisk evil_gordita (robert@ip70-188-56-139.rn.hr.cox.net) |
| 22:58.23 | *** join/#asterisk theron (~theron@173-20-126-202.client.mchsi.com) |
| 23:09.24 | *** join/#asterisk robink (~quassel@unaffilated/robink) |
| 23:10.30 | robink | Is it appropriate to ask SIP questions in here? |
| 23:14.30 | mjordan | robink: you can, although it is a Sunday evening, so not a lot of folks are paying attention. |
| 23:15.22 | robink | mjordan: OK. Just as well, probably, as the question is rather asinine and one that a. is likely being caused by deficiencies in a SIP client, and b. would probably be solved by handling the call setup (as it should be, as this is with a SIP trunking account) |
| 23:15.46 | robink | I'm having difficulties associating an outbound call with a DID |
| 23:15.59 | robink | The call completes, but does not show as coming from my inbound calling number |
| 23:16.40 | robink | There is no way to associate an account or match/routing with an outbound DID, so I'm assuming the client is expected to send something in its SIP INVITE statement to the POTS trunk. |
| 23:17.22 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 23:18.02 | robink | It's too bad I can't get the SIP client in question to spit out its SIP statments |
| 23:19.54 | phix | mjordan: it is a monday morning! |
| 23:20.44 | phix | robink: you can use the asterisk console if you hook the sip client to an asterisk server |
| 23:20.53 | phix | and see what it does |
| 23:21.03 | robink | phix: Yep, which is what I should have started with |
| 23:21.24 | phix | :D |
| 23:21.32 | phix | Then your questions would be within topic too |
| 23:21.47 | robink | phix: I have an embedix box essentially dedicated to VoIP (and other light-duty tasks), but I've yet to fire up Asterisk on it, despite the fact that this is a SIP trunking account, not a residential plan. |
| 23:22.50 | phix | ah, Asterisk is great at SIP trunking. I have set up a few boxes for clients that does that |
| 23:42.08 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-qvecpniglwsxdign) |
