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| 00:40.00 | Samot | Well that broken exhaust hose on the AC unit might explain why it's been Hotlanta like in the house. |
| 00:42.29 | Samot | drmessano: I can understand why you're bald. Hair just has no place in conditions like that. |
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| 05:47.48 | _abc_ | Hello. Can anyone please tell how one can use the linux kernel's built in echo canceller device in asterisk? I read it can be used, but how? Recompile asterisk? Is there a howto? |
| 05:48.35 | _abc_ | This is for non pstn voip to voip only service |
| 05:51.36 | _abc_ | Only really pertinent info I could find is at http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation |
| 05:57.07 | _abc_ | Okay, I understand why there is silence on official asterisk channels about the echo canceller issue, it is a competing product although gpl. Point taken. |
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| 07:20.10 | joj[] | nIs it possible to read pjsip configuration objects of type=wizard using realtime (yet)? Doesn't look like it to me since there is no matching object type mentioned in socery.conf.sample? |
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| 07:51.45 | slima | Hi, I wonder how to do something like that: Dial -> 1 (if busy -> dial -> 2 (if busy - > dial -> 3)) |
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| 09:22.12 | ChannelZ | slima, you can check ${DIALSTATUS} and see if it reported CONGESTION and behave accordingly |
| 09:22.45 | ChannelZ | (and/or BUSY) |
| 09:27.24 | slima | ChannelZ: thx for the hint. |
| 09:32.09 | ChannelZ | Sure |
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| 10:19.19 | orn | Is anyone here using FollowMe? If so, why? I can't figure out what the benefits of using FollowMe over just using the dialplan are. |
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| 12:12.29 | slima | I want somthing like that: Dial -> 1 (if busy -> dial -> 2 (if busy - > dial -> 3)) I did http://pastebin.com/J2qP50dL but this don't work, why? |
| 12:13.47 | slima | call-limit=1, 02 and 03 talking, 04 is onlne and idle, but *** auto4 ${DIALSTATUS} *** says: *** auto4 CHANUNAVAIL *** |
| 12:16.48 | rbeer | Do you have the output from the cli? |
| 12:22.59 | Gugge | slima: isnt that what you expect ... the auto4 status is from the Dial to 03 |
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| 12:27.33 | slima | hmm, so what can I do? |
| 12:28.26 | Gugge | change the dialplan logic to do what you want |
| 12:28.59 | slima | very helpful |
| 12:29.11 | Gugge | right now, you dial 1 |
| 12:29.18 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 12:29.25 | Gugge | if dialstatus is "CHANUNAVAIL" you go to auto3 |
| 12:29.28 | Gugge | and dont dial 02 |
| 12:30.02 | Gugge | auto3 outputs the dialstatus again, and then goto auto4 |
| 12:30.19 | Gugge | and that skips the dial to 04, and goto auto5 |
| 12:33.21 | *** join/#asterisk guest9103 (~bounceman@185.32.9.250) |
| 12:33.41 | guest9103 | Hello, I want to see what is going on inside chan_sip module during a call, what debug options are available? |
| 12:46.21 | _bittis_ | hi everyone, quick question |
| 12:46.26 | Samot | Ugh. I hate working with Indians. |
| 12:46.47 | _bittis_ | do you know if all version of Lua are compatibily with asterisk? |
| 12:46.57 | _bittis_ | Samot: Little Endians? |
| 12:47.15 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:47.29 | Samot | They just can't think for themselves. |
| 12:47.58 | _bittis_ | most people can not |
| 12:48.07 | Samot | And wont admit when they can't do something. |
| 12:48.13 | guest9103 | Please come again |
| 12:48.42 | Samot | They designed a softphone app for a client we share. |
| 12:48.50 | Samot | They say presence works on their app. |
| 12:49.22 | Samot | But they've never gotten the presence server to work. |
| 12:49.33 | Samot | So how they got it to work on their app?!! |
| 12:50.02 | _bittis_ | some sales guy probably to blame here :P |
| 12:50.06 | Samot | It just meant that their app changes statuses in the gui. |
| 12:50.37 | Samot | So the app will show you Away but since there's no backend to send it to, pfffft. |
| 12:54.21 | Samot | I just joined two weeks ago but they have zero tech docs or any docs on what they've actually done, what's been tested, etc. They're telling the client it's 90% ready and he's about to spend $6K+ in colocation, equipment, etc to expand for this service and the reality is, it's really about 45% done. |
| 12:57.05 | [TK]D-Fender | That's because it's the project is 50% BS and considered fulfilled in real-time ;) |
| 12:59.11 | Samot | No kidding. |
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| 13:02.08 | Samot | They are insistent that the app is a 100% working and that any flaws are in the voice system backend. But how can you say your app is a 100% working if the features it connects to on the voice server never functioned?!! |
| 13:11.07 | orn | What's that got to do with them being Indians? Sounds like a problem you'd get with a shitty software company, rather than anything to do with them being Indians. You get what you pay for :-) |
| 13:11.48 | _bittis_ | don't outsource, do everything inhouse |
| 13:12.23 | makowals | Hi, when dialing "PJSIP/12345@tunnel" and having two aors assigned to endpoint "tunnel" where the first one is Unavail and the second one Avail, is it expected behaviour for the call to trying to pass through the Unavail one (because it's the first one) anyway? |
| 13:13.07 | makowals | Shouldn't it be automatically directed to the available one if multiple aors are assigned to one endpoint? |
| 13:20.25 | Samot | orn: It's because of experience with India. It's just their culture. |
| 13:20.49 | Samot | orn: They are still a caste society, which means if you are considered the "boss" they won't do anything you don't tell them. |
| 13:21.09 | Samot | orn: And they never admit they can't do something or delivery something. |
| 13:21.50 | Samot | To the point that if you say "Can you do this?" they will say yes but the reality is, they've never done it. |
| 13:22.47 | Haris | can asterisk cdr do call recording to webrtc calls ? |
| 13:22.54 | Haris | ws/wss video calls ? |
| 13:23.16 | Haris | call recording to webrtc calls = call recording of+ webrtc calls |
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| 13:33.16 | [TK]D-Fender | <Haris> can asterisk cdr do call recording to webrtc calls ? <- HUH? |
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| 13:41.13 | paulgrmn | Hi. I have Yealink phones on a NAT'd LAN talking to Asterisk 13.8.2 running on Amazon EC2. The call history on the phones gets the private address of the EC2 instance rather than its public IP, so call backs don't work. 'external_media_address' and 'external_signaling_address' are set to the server's public IP for the PJSIP transport. Any ideas? Thanks. |
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| 14:14.33 | Samot | paulgrmn: So Asterisk knows that not know the endpoint is behind NAT but that it is also behind NAT? |
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| 14:49.38 | paulgrmn | Well, calling works, so Asterisk seems to deal well enough with the NAT in every other respect. I set 'local_net' for the transport with a value that encompasses the server's private IP address; if I omit that, calling breaks in one direction. |
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| 15:07.04 | paulgrmn | Looking at a packet capture, the only obvious place I see this being sent is in the dialog-info XML for NOTIFY's, like: <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="3" state="full" entity="sip:126@172.nn.nnn.200:5060">. The SIP headers (To, From, Contact) show the correct public IP. |
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| 17:48.49 | drmessano | Wow, RIP Polycom |
| 17:51.57 | drmessano | Private Equity Firm = Slash and Burn to minimize losses |
| 18:08.05 | *** join/#asterisk sergiobaiao (~sergio@189-81-117-142.user.veloxzone.com.br) |
| 18:10.06 | sergiobaiao | hello there |
| 18:11.21 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 18:11.42 | sergiobaiao | can i get some advice here? i'm willing to create a voip system to allow me to use my iphone as a softphone. This voip system would dial out over a pstn landline and also over sip provider. can you advice on a fxo adapter with one port only that works well with asterisk? |
| 18:12.11 | Mango45 | sergiobaiao, OBi110. |
| 18:12.44 | drmessano | SPA-3102 |
| 18:13.39 | Mango45 | The SPA-3102 is great if you like 10-year-old technology that sounds like crap. |
| 18:13.47 | sergiobaiao | i tried with an ht-503 but couldn't make it to work |
| 18:14.26 | Mango45 | Why not? |
| 18:15.45 | sergiobaiao | when i set the trunk to authenticate, it won't get the line. if i set it to noauth, it shows exception 'Exception' with message 'PBX_Asterisk_Interface_SIP_NoAuth dont support channel returning, |
| 18:17.53 | Mango45 | I am not familiar with that error but perhaps another member here is. |
| 18:18.02 | Samot | Wait a minute. |
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| 18:18.28 | Samot | The one time drmessano actually recommends decade old equipment and he's crapped on. |
| 18:18.43 | Samot | Normally he's a jerk for telling people to upgrade. |
| 18:18.48 | Samot | Dude, you can't win. |
| 18:18.54 | drmessano | I can't |
| 18:18.58 | drmessano | Howevwr |
| 18:19.02 | drmessano | However* |
| 18:19.24 | drmessano | I know how to make an SPA-3102 work and not sound like crap |
| 18:19.36 | Samot | This is true. |
| 18:20.10 | Samot | That SPA3102 you helped with, rock solid. No complaints about anything. |
| 18:20.39 | Mango45 | Interesting; most people report that it has unsolvable echo. |
| 18:20.43 | drmessano | When you know what you're doing, they work and work fine |
| 18:20.52 | Mango45 | Are you using it on an analog line? |
| 18:20.57 | Samot | Yup. |
| 18:21.04 | drmessano | Thats kind of its purpose |
| 18:21.09 | Samot | This is doing PSTN toVOIP and vice versa. |
| 18:21.19 | drmessano | What else would you use it on? |
| 18:21.26 | sergiobaiao | the HT-503 seems like to not work as a gateway |
| 18:21.33 | Samot | Rings a cordless phone for a few seconds then transfers to the PBX. |
| 18:21.52 | Mango45 | Cable VoIP. |
| 18:22.02 | sergiobaiao | it looks like we have to add its FXO port as an extension, than call that extension before calling external |
| 18:22.23 | drmessano | Mango45: That's still an analog line. Maybe you're referring to something that's copper for much more than a few feet of its life |
| 18:23.23 | drmessano | Even still the conversion to Analog puts it in the same domain, whether its coming from the SLIC or the CO |
| 18:23.33 | drmessano | or another ATA |
| 18:23.48 | drmessano | Regardless, the SPA-3102 works fine |
| 18:24.13 | drmessano | I've actually used it with a providers ATA and on some awful copper lines from the CO |
| 18:24.30 | Mango45 | Ok. Are you using it on something that's copper for much more than a few feet of its life? |
| 18:24.40 | sergiobaiao | drmessano, what should be done on it to work flawless? here in Brazil its easier to find the SPA3102 than the Obi110 |
| 18:24.46 | drmessano | I think I just said that |
| 18:25.12 | drmessano | sergiobaiao: Match the levels and line impedence. That's 99% of the battle. |
| 18:25.25 | Mango45 | How do you know the line impedence? |
| 18:25.26 | drmessano | Use what's default for telco lines there, then tweak.. |
| 18:25.50 | drmessano | Mango45: Go with what is published for your country, then tweak from there |
| 18:25.51 | Samot | Because things work off standardization |
| 18:25.56 | sergiobaiao | Mango45, there's a default for almost every country |
| 18:26.01 | Mango45 | Ah. |
| 18:26.06 | sergiobaiao | here in Brazil i thing is 900 |
| 18:26.50 | drmessano | http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/spa3102-voice-gateway-router/108731-pqa-108731.html |
| 18:26.59 | Samot | lol "i thing" |
| 18:27.08 | drmessano | Literally #4 |
| 18:27.11 | drmessano | Set the FXO port Impedance on the PSTN tab to 220+820||120nF, and set FXS Port Impedance to 220+820||115nF as a starting point. |
| 18:27.22 | drmessano | Thats for the USA |
| 18:27.27 | Samot | And that is actually not a poke at you sergiobaiao. |
| 18:27.41 | sergiobaiao | oops, here is 600 |
| 18:28.29 | sergiobaiao | mistyped |
| 18:29.01 | Samot | I know but "i thing" it's just an old joke from some friends of mine back in the day. |
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| 18:29.28 | Mango45 | Anyone happen to know what the impedance is for Canada? |
| 18:29.29 | Mango45 | Googles |
| 18:29.46 | drmessano | sergiobaiao: 600 under perfect conditions.. It very likely has some inductive reactance |
| 18:29.55 | Samot | Canada and the US have a standardized system. |
| 18:30.04 | Mango45 | Thanks. |
| 18:30.40 | Samot | Thank AT&T. They did it. |
| 18:31.14 | Samot | Or Bell..whatever they called themselves in the 40's or 50s. |
| 18:31.42 | sergiobaiao | drmessano sure |
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| 18:34.30 | drmessano | The SPA-3102 does not work purely resistive 600 ohms on USA/CA land lines |
| 18:34.30 | Mango45 | Looks like 600 is the default on both the SPA3102 and OBi110. |
| 18:34.37 | drmessano | Yeah |
| 18:34.42 | sergiobaiao | i was searching here for obihai's adapters |
| 18:34.44 | drmessano | and thats really not correct |
| 18:35.01 | sergiobaiao | i found obi110, obi200 and obi300 |
| 18:35.04 | drmessano | Ive NEVER ran into a purely resistive 600 ohm line from a telco, ever |
| 18:35.19 | sergiobaiao | and the price difference between them is minimal |
| 18:35.52 | Samot | That's because Cisco wanted telco's to use their UCM |
| 18:36.01 | drmessano | sergiobaiao: http://www.2l2o.com/open/spa-3102 |
| 18:36.09 | drmessano | I wrote that.. Feel free to use it |
| 18:36.16 | Samot | It works. |
| 18:36.28 | Samot | It even worked on an SPA3102 that's in the UK. |
| 18:37.08 | drmessano | I found a page showing Brazil is 600 ohms as well. so the 220+820||120nF should be the same there too |
| 18:37.21 | sergiobaiao | drmessano, thanks for that |
| 18:38.18 | drmessano | If you end up with Echo, you didn't follow the guide |
| 18:38.26 | drmessano | Including the RESET part |
| 18:38.39 | sergiobaiao | ;) |
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| 18:38.45 | drmessano | Ask Samot |
| 18:39.00 | drmessano | Because him and I are buds.. and I gave him that guide |
| 18:39.08 | drmessano | and he told me it didnt work |
| 18:39.13 | drmessano | Because he didnt RESET |
| 18:39.28 | drmessano | So now he gets a reminder |
| 18:39.35 | drmessano | FOREVER |
| 18:39.39 | sergiobaiao | LOL |
| 18:39.41 | Samot | Whatevs. |
| 18:39.51 | drmessano | hahaha :) |
| 18:40.01 | Samot | I can still change the wheels on a hand truck. |
| 18:40.01 | drmessano | It was kinda funny actuall |
| 18:40.03 | drmessano | It was kinda funny actually |
| 18:40.34 | drmessano | Wasnt it like "DID you RESET IT?" |
| 18:40.38 | drmessano | "YES, wait.. no" |
| 18:40.52 | Samot | I said I followed all the instructions. |
| 18:40.54 | Samot | You asked that |
| 18:40.57 | Samot | and I said... |
| 18:41.01 | Samot | "Well except for that ONE" |
| 18:41.01 | drmessano | Oh right |
| 18:41.15 | drmessano | Thats even better, actually |
| 18:41.20 | Mango45 | Newbie here. If the impedance is 600 ohms here, why do we choose 220+820||120nF? |
| 18:41.26 | Samot | Because I thought I was too cool for school and didn't need to reset. |
| 18:41.39 | drmessano | Mango45: Because 600 ohms exists in a lab, not on a real line |
| 18:41.57 | drmessano | Mango45: Copper wire, lots of distance, things affect impedence |
| 18:42.07 | drmessano | Twists, turns, bends, splices |
| 18:42.11 | Samot | Crapping connectors. |
| 18:42.13 | sergiobaiao | there's a lot of resistance over the line |
| 18:42.17 | drmessano | At the end, it's more inductive reactive |
| 18:42.55 | Samot | Poor management of remote boxes.. |
| 18:43.07 | Samot | That have rain and whino urine soaking the wires. |
| 18:43.11 | Mango45 | Very interesting. So if I were to connect an ATA to a fiber ONT with a short copper loop, I would choose 600? |
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| 18:43.40 | drmessano | Mango45: MAYBE.. But in theory, it should be 600 ohms |
| 18:43.55 | drmessano | Even still |
| 18:44.35 | drmessano | NOBODY tests equipment like that.. it's not manufactured for 600 ohms.. There's a line coupling transformer inside the box that supposedly is 600 ohms |
| 18:45.00 | drmessano | But even the 4 inches from the transformer to the screw terminals is an impedence bump |
| 18:46.07 | drmessano | At the end of the day, the spacing on all the terminals, the twists in the wires, everything is designed for 600 ohms.. when its actually hung, who knows .. |
| 18:46.21 | Samot | Yup. |
| 18:46.32 | Samot | Not to mention.. |
| 18:46.40 | Samot | Have you seen a CO or a junction box? |
| 18:46.59 | Samot | You'd think the Telco would be the best looking wiring you've ever seen. |
| 18:47.11 | freebs | u 2 calm down |
| 18:47.37 | drmessano | RF is far more sensitive to impedence changes.. and if you go buy what it takes to keep the impedence solid for RF, good luck with telco stuff |
| 18:47.43 | Samot | Nope, it's been wired by a guy who's working 12 hours already and has to stop to get your DSL working, so he probably twisted your line off somewhere and didn't even cap it. |
| 18:48.06 | drmessano | You cant put open wire feeders 12 inches from a structure, but people staple phone lines to everything |
| 18:48.14 | drmessano | So, 600 ohms? HA |
| 18:48.22 | Mango45 | So can your average person (i.e. me) test this scientifically or is it just a matter of experimenting and seeing what "sounds good"? |
| 18:48.52 | drmessano | Mango45: Its not practical to test. Start at where I showed, then go each direction.. |
| 18:49.19 | Mango45 | Thanks; I learned a lot. |
| 18:49.39 | Samot | People who spend time testing that never moved out of their parents house or knew the scent of a woman that didn't require a payment after. |
| 18:49.47 | Mango45 | omg |
| 18:50.14 | drmessano | Since the settings are limited anyway, you dont have many options.. just start with 220+820||120nF, then I think there's 115nF and a few other options |
| 18:50.20 | sergiobaiao | wow, that's low |
| 18:50.33 | drmessano | lol |
| 18:50.54 | Samot | No, it was funny. |
| 18:50.55 | Mango45 | on 110 the setting is: |
| 18:51.18 | Mango45 | 220+(820||120nF) and 220+(820|115nF) |
| 18:51.31 | drmessano | Yeah |
| 18:52.06 | Mango45 | In any case, I stand corrected. drmessano knows what he is talking about. :) |
| 18:52.27 | drmessano | lol |
| 18:52.32 | Samot | Some would argue that. |
| 18:52.44 | drmessano | Yeah, my ex wives |
| 18:52.51 | drmessano | But they are in a better place now |
| 18:52.53 | drmessano | All of them |
| 18:53.10 | drmessano | Better Place = 2 states away or more |
| 18:53.45 | Samot | Yeah but someone will find the graves. |
| 18:53.56 | Samot | I'm sure they're shallow. |
| 18:54.02 | Mango45 | tests his line |
| 18:54.32 | Mango45 | wait |
| 18:54.38 | Mango45 | I have no one that I need to talk to :( |
| 18:56.34 | drmessano | Mango45: I would also make the argument that while the SPA-3012 is 10+ years old, I have no reason to think anything newer would come along to better handle analog lines. Even Cisco didn't replace it when they bought Linksys, because it worked. That holds a lot of water. |
| 18:56.49 | Samot | Yeah they did. |
| 18:56.50 | drmessano | It even does T.38 |
| 18:57.01 | ruel | Anybody ever seen a SIP call being sent in from a provider about 8 times simultaneously? (From a single incoming call.) ? |
| 18:57.04 | Samot | Like the 232D or whatever it's called. |
| 18:57.38 | ruel | My server seems to be responding properly. But I want to be sure it's not my fault... |
| 18:57.42 | Samot | How do you know it's a single incoming call? |
| 18:57.57 | Mango45 | ruel, yes, when my reply to their INVITE gets blocked by a firewall. Did you check their CDRs to be sure some telemarketer did not literally call you 8 times at once? I've seen that happen too. |
| 18:58.09 | drmessano | Show us on the debug doll where the call touched you |
| 18:58.51 | Mango45 | drmessano, as long as you can make it sound good, that is fine. If I were buying a new one today, I would buy an OBi, in part because of its excellent and powerful call routing features not present in the SPA. |
| 18:59.31 | drmessano | I have Asterisk for that :) |
| 18:59.33 | ruel | Mango45: Yes I checked their CDRs and they have SIP trace too. I don't have Asterisk debug cuz it has only happened about 4 times in the past 2 days at random. Wasn't telemarketers, was a regular person. |
| 18:59.44 | drmessano | I just want my dumb box to interface |
| 19:00.09 | sergiobaiao | Well, i've learned some from you today. for that, thank you guys. i'll be back after my ATA arrives ;) |
| 19:00.14 | sergiobaiao | see you when i see you |
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| 19:01.06 | drmessano | Thats actually why I wrote that guide.. Because too many people had the setup of the SPA3102 WRONG. Using the SPA dialplan for routing calls |
| 19:02.06 | drmessano | Granted, you still do.. but it's not flexible.. They used fixed extensions and other BS |
| 19:02.23 | drmessano | This just takes the call, hands it off.. |
| 19:03.47 | ruel | Mango45: if my server is responding to their INVITE though according to their SIP trace, it wouldn't be a firewall issue would it? |
| 19:04.48 | ruel | They send an INVITE, Asterisk sends back a TRYING, then an OK, and then one of them sends a BYE a few seconds later (4-6 seconds). |
| 19:05.01 | ruel | Some calls Asterisk sent the BYE and some they did first. |
| 19:05.10 | Mango45 | Correct, if their SIP trace has your response, then my guess is not correct. |
| 19:05.14 | ruel | Then after about 8 times, it goes through properly. |
| 19:05.38 | Mango45 | You might want to post a SIP trace (personal info sanitized) for the experts here to look at. |
| 19:06.02 | ruel | k will do that later. Have to fry other fish for now. Thanks for the input. |
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| 19:07.07 | drmessano | What Mango45 said.. Post some debug. Describing the call tends to make a lot of us stabby |
| 19:07.23 | drmessano | Its like telling us how your engine sounds rather than bringing it around |
| 19:09.20 | ruel | Yes I will. Just wanted a bit of input/ideas so I could research a bit more thoroughly before bringing someone else into it. Thanks |
| 19:09.44 | drmessano | No worries |
| 19:10.27 | Alex_Bkash | ruel |
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| 21:08.24 | jpsharp | Is there a way to set a group on a channel via AMI originate? |
| 21:08.37 | jpsharp | b |
| 21:14.21 | jpsharp | Seems I can only do it via Dial, not Originate. |
| 21:15.45 | WIMPy | If yu're lucky a simple SetVar will do it. Otherwise use the dialplan. |
| 21:17.35 | jpsharp | I've tried setting the OUTBOUND_GROUP var in the Originate Variable: command, but that doesn't seem to do it. And perusing the code, it appears that that var is only ever references in app_dial.c |
| 21:17.49 | jpsharp | So it seems I have to add it somewhere else. |
| 21:18.18 | WIMPy | Are you asking about a group or some variable of yours? |
| 21:23.51 | jpsharp | Group |
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