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| 00:13.33 | WIMPy | wastonNope: I hear what you see. Seems to happen if you don't hang up for long enough. |
| 00:13.59 | wastonNope | hmm.. hang up for long enough |
| 00:14.10 | wastonNope | then asterisk spits some caller-id at you? |
| 00:14.28 | wastonNope | the caller-id is composed of the extension number and name |
| 00:14.40 | WIMPy | No idea. I don't have a POTS phone that supports caller ID. But I get a short ring. |
| 00:14.56 | wastonNope | hmm |
| 00:15.23 | WIMPy | The last time I used an analogue phone was before caller ID was invented. |
| 00:17.06 | wastonNope | maybe this is a sign that I should get some wifi sip phones? |
| 00:17.28 | WIMPy | No, you don't want them. |
| 00:17.30 | wastonNope | I have this aversion to spending money - I think I'm allergic to it |
| 00:17.59 | WIMPy | tries to find the right configuration to be able to dial... |
| 00:18.12 | WIMPy | :-) |
| 00:18.55 | wastonNope | I don't want wifi sip? |
| 00:19.12 | wastonNope | I'm able to dial and receive calls, no problem |
| 00:19.15 | WIMPy | No, you don;t want to use wifi for realtime. |
| 00:19.51 | wastonNope | everything seems to work as it should - except for this caller-id blast at the end of calls or when I restart asterisk, or update the config in freepbx |
| 00:20.41 | wastonNope | what product do you recommend for walking around the house and talking to people that have called me? |
| 00:21.01 | WIMPy | A normal DECT / CAT-iq phone. |
| 00:21.13 | WIMPy | You can get a SIP base station if you like. |
| 00:21.27 | wastonNope | oh - digital to the base station |
| 00:21.36 | wastonNope | DECT to the face |
| 00:21.37 | WIMPy | Digital all the way. |
| 00:22.32 | wastonNope | yup - that's what I'll do - thanks, WIMPy - you seem to be fairly interactive despite your name |
| 00:23.05 | wastonNope | I'd still like to figure out what the heck is going on with this caller-id blast, though - for my own curiosity and for the health of the project |
| 00:23.08 | WIMPy | Well, I wish I could say the same about Asterisk/DAHDI. |
| 00:23.13 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 00:23.47 | WIMPy | You could try to disable call waiting/hold / 3-way calling. |
| 00:24.17 | wastonNope | you said something about the hang up length - but I receive this caller-id blast down the line even if a call hasn't been made - if I restart asterisk for instance |
| 00:24.31 | wastonNope | I will try |
| 00:24.44 | WIMPy | I don't see (hear) that. |
| 00:24.58 | wastonNope | the caller-id has the name and extension number - I'm thinking there is some protocol that uses caller-id to tell analog phones "who" they are |
| 00:25.16 | wastonNope | you know, some analog phone that has a digital display that shows its name and ext number |
| 00:25.27 | WIMPy | Hmmm/ |
| 00:25.36 | WIMPy | There was that thing called adsi. |
| 00:25.36 | wastonNope | I |
| 00:25.55 | wastonNope | I'll record a caller-id blast on a dead line.. give me a min |
| 00:27.22 | wastonNope | that sounds like an interesting path of research into this problem.... |
| 00:29.02 | WIMPy | Is it possible to have an invalid extension with immediate=no? |
| 00:30.50 | wastonNope | not sure- this is day 1 for me |
| 00:33.33 | wastonNope | WIMPy: https://soundcloud.com/user378628101/asterisk-caller-id-blast-on |
| 00:34.22 | WIMPy | I have no idea what I could make of that. |
| 00:35.35 | wastonNope | all I did was make a change to a description field in FreePBX, then hit "Apply Config" - I'm assuming it is restarting some service - in doing so, I get that caller-id blast down the analog line |
| 00:36.06 | wastonNope | I'm guessing to update the LCD display on some analog piece of equipment with the name and ext number |
| 00:36.27 | wastonNope | but my equipment don't play like that |
| 00:37.16 | WIMPy | Shit. At first I thought, immediate=no solves the dundi issue, but it doesn't. |
| 00:37.18 | WIMPy | Owh well. |
| 00:38.04 | wastonNope | dundi? |
| 00:38.45 | WIMPy | ~dundi |
| 00:38.45 | infobot | well, dundi is at http://www.dundi.com. DUNDi, an optional Asterisk component, is a distributed, decentralized peer to peer network that provides routes to PSTNs between peers on the same DUNDi network. |
| 00:41.26 | wastonNope | cool - this mark spencer guy needs to get a hobby ;) |
| 00:41.37 | wastonNope | making all this cool software |
| 00:42.51 | wastonNope | years ago I used to host asterisk.org for him |
| 00:42.55 | WIMPy | Unfortunatly (almost) noone seems to be interested in the good bits. |
| 00:45.02 | [TK]D-Fender | [19:28]WIMPyIs it possible to have an invalid extension with immediate=no? That should force a call against "s", no? |
| 00:45.51 | WIMPy | [TK]D-Fender: Well, it doesn't make sense to use no anyway. That just seemed to work better by chance. |
| 00:46.09 | wastonNope | [TK]D-Fender: https://soundcloud.com/user378628101/asterisk-caller-id-blast-on |
| 00:46.36 | wastonNope | another caller-id blast down the analog line without any use of the circuit |
| 00:46.48 | [TK]D-Fender | wastonNope: heard it earlier.. and I'm not sure why that is happening... Post up configs, etc on the tracker in a bug report and see what comes back. |
| 00:47.18 | wastonNope | ok - second recording is just dead space - then caller-id blip - then more dead space |
| 00:47.20 | wastonNope | will do |
| 00:59.07 | WIMPy | Is there no option to completely disable hookflash support? |
| 01:57.23 | ipengineer | When using the asterisk REALTIME cmd is there a way to match on multiple columns? |
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| 04:24.00 | wacomcito | hi all |
| 04:25.34 | wacomcito | i have a problem with auth sip trunk. I think the problem is 2 clients at same IP with different user. |
| 04:25.42 | *** join/#asterisk Katty (~Kattytoo@24-207-199-117.dhcp.stls.mo.charter.com) |
| 04:25.56 | wacomcito | WARNING[13869][C-0000002d]: chan_sip.c:16326 check_auth: username mismatch, have <RTB_casa>, digest has <pstn-fix> |
| 04:25.59 | Katty | ohai. |
| 04:26.18 | wacomcito | any help? |
| 04:26.32 | Katty | i hear chocolate helps. |
| 04:26.38 | Katty | maybe some godiva liquor? |
| 04:27.00 | saint_ | 2 clients at same ip with same diff users does not matter |
| 04:34.14 | Katty | herp derp |
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| 04:57.10 | wacomcito | solved :) |
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| 05:32.15 | boratynskikamil | Good morning. |
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| 07:04.12 | illuder | hello all, noob here on IRC..tx. |
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| 07:11.10 | ChannelZ | points |
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| 10:02.21 | boratynskikamil | Is it better recommended to write SIP users in users.conf or sip.conf? |
| 10:03.06 | WIMPy | users.conf is best used to practice the use of rm. |
| 10:14.34 | boratynskikamil | WIMPy: As I thought. :-) |
| 10:15.14 | boratynskikamil | And the extensions file? .conf or AEL? For me it does not matter, but as far as I may consider. AEL is newer then conf. |
| 10:16.20 | WIMPy | Well, AEL is just parsed to regular conf anyway. |
| 10:16.40 | boratynskikamil | WIMPy: Really? |
| 10:16.58 | WIMPy | It may look nicer to some, but the standard conf will give you best support. |
| 10:17.22 | WIMPy | Yes, it's more like a preprocessor thing. |
| 10:17.31 | boratynskikamil | WIMPy: Yeah, yeah. |
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| 15:17.48 | wastonNope | http://www.youtube.com/watch?v=bPXVGQnJm0w |
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| 18:23.46 | wastonNope | I would like to set fxs_immediate=yes on only one fxs channel, not all of them. I'm using the current stable AsteriskNOW distribution. I'm at a loss as to what config file to place this in. Any help would be appreciated. |
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| 18:26.26 | ChannelZ | I don't even know what that is. There is just an 'immediate' in chan_dahdi.conf |
| 18:27.14 | wastonNope | I don't think that is the same |
| 18:27.22 | wastonNope | let me find what I was reading... |
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| 18:28.29 | wastonNope | I believe that immediate doesn't wait to see what digits are dialed and connects immediatly to an extension. Say for something like a Batphone |
| 18:28.59 | ChannelZ | and what is it you are trying to make happen? |
| 18:29.03 | wastonNope | fxs_immediate=yes immediatly answers a ringing FSX line and doesn't wait for the caller-id information |
| 18:29.23 | wastonNope | I have an FSX port hooked up to a doorbell that rings a phone line |
| 18:29.38 | wastonNope | right now, I'm picking up the inbound route and sending to a queue |
| 18:29.59 | wastonNope | however, it rings a few times before being answered as it is waiting to hear the caller-id announcement |
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| 18:30.13 | wastonNope | there is no caller-id, so it just needs to answer right away |
| 18:30.38 | wastonNope | which fxs_immediate=yes will do as I understand it - but I don't want to enable it for all fxs ports, just port 3 |
| 18:30.43 | ChannelZ | so turn off caller ID for that channel |
| 18:30.50 | wastonNope | okay |
| 18:31.05 | wastonNope | same problem - where do I put that? callerid=no |
| 18:31.08 | ChannelZ | but in either case, you set the option before the channel => specification to set it for that channel. |
| 18:31.15 | ChannelZ | Then set it back to whatever prior to the rest. |
| 18:31.17 | WIMPy | There's only an immediate without fxs_. |
| 18:31.26 | wastonNope | usecallerid=no |
| 18:31.47 | WIMPy | But immediate probably doesn't make any sense on fxo. |
| 18:31.51 | wastonNope | ChannelZ: I see |
| 18:32.44 | wastonNope | WIMPy: I see /usr/dadhi/genconf_parameters:#fxs_immediateyes |
| 18:33.35 | ChannelZ | options in chan_dahdi are basically sequential, you set up all the options and then when it hits a channel => line it assigns those paramaters to the channel(s) being defined. |
| 18:33.39 | WIMPy | Well, that's just for genconf. |
| 18:33.54 | wastonNope | ChannelZ: thanks! |
| 18:34.36 | wastonNope | I'm wondering if my modification of /etc/asterisk/chan_dahdi_groups.conf will be overwritten by FreePBX |
| 18:35.36 | ChannelZ | chan_dahdi.conf is the *actual* file being read, so things are parsed sequentially. If they #include other files, it will happen 'in order' where that statement is. |
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| 18:36.04 | ChannelZ | So if the channels get defined somewhere by FreePBX before your manual ones, they win. |
| 18:37.09 | wastonNope | ChannelZ: it worked like a charm |
| 18:37.58 | ChannelZ | turning off CallerID? |
| 18:38.00 | wastonNope | I'm guessing I should put this in a seperate config file that is included by chan_dahdi.conf |
| 18:38.01 | wastonNope | ChannelZ: yes |
| 18:38.35 | ChannelZ | I thought FreePBX generally put "### THIS FILE WILL BE OVERWRITTEN ###" type comments in the config files it pisses all over |
| 18:38.39 | wastonNope | can channels be referenced more than once? |
| 18:38.52 | WIMPy | no |
| 18:38.57 | ChannelZ | No, the channels => line "consumes" the channels |
| 18:38.57 | wastonNope | ChannelZ: yes, this file has that |
| 18:39.29 | ChannelZ | well then #including your own from there won't matter either since your #include statement will get written over. |
| 18:39.33 | wastonNope | okay - so how do I specify custom channel parameters, such as turning caller-id off and not have it overwritten by FreePBX? |
| 18:39.35 | ChannelZ | This is why I hate GUIs |
| 18:40.04 | WIMPy | And that's why you have to ask that question in #freepbx. |
| 18:40.32 | wastonNope | it looks like it provides #include chan_dahdi_general_custom.conf for custom config enteries |
| 18:40.42 | ChannelZ | write protect it and change the permissions so freepbx can't write to it :) (unless it runs as root) |
| 18:40.46 | wastonNope | but I'm not sure how I can reference a channel twice |
| 18:40.51 | wastonNope | I'll ask in #freepbx |
| 18:41.10 | wastonNope | ChannelZ: ha! that doesn't sound very nice to freepbx |
| 18:41.24 | WIMPy | Yes, write protection usually works. But you never know what happens if it fails to write. |
| 18:41.32 | wastonNope | exactly |
| 18:43.38 | wastonNope | ChannelZ: WIMPy and I were looking at another problem yesterday - on my FXO port I have an analog phone with caller-id. After every call or asterisk re-hup I get a caller-id blast. For example, if I take the phone off hook for a few seconds, then hang up, a few seconds later I get a caller-id blast with the extension name and number |
| 18:43.49 | wastonNope | the problem is that my analog phone registers this as a missed call |
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| 18:45.29 | wastonNope | ChannelZ: here is what comes down the line during an asterisk reload: https://soundcloud.com/user378628101/asterisk-caller-id-blast-on |
| 18:48.04 | WIMPy | And I'd like to completely disable multiple calls. But can only find the callwaiting parameter. |
| 18:55.52 | ChannelZ | I don't know why asterisk would be doing that |
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| 18:59.55 | WIMPy | Oh, and I'd also like to know why I get a ringback tone instead of a dialtone and why it doesn't stop when dialling. I'm very sure that didn't happen last time I tried dahdi. |
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| 19:06.22 | JoshBon | Hello - I was speaking on here the other day about an issue with the network on a PBX and SIP provider. On an MTR report I get a loss of 90%+ but the packets continue to go through without issue, the voice quality is awful - Would this be the cause? |
| 19:07.53 | ChannelZ | Well packet loss is certainly not going to make for good calls. |
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| 19:08.38 | wastonNope | hmm... where can I find documentation on chan_dahdi.conf ? |
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| 19:09.31 | ChannelZ | it's all pretty much in the stock chan_dahdi.conf |
| 19:09.38 | WIMPy | Only in chan_dahdi.conf.sample |
| 19:09.42 | [TK]D-Fender | wastonNope: The sample config |
| 19:10.10 | wastonNope | [TK]D-Fender: right, duh |
| 19:10.29 | JoshBon | What is the loss meaning exactly? |
| 19:10.40 | JoshBon | As the packets sent on the before, current and after count all match |
| 19:10.44 | wastonNope | foobar - I don't have a sample config in my distro |
| 19:10.56 | wastonNope | I found http://www.voip-info.org/wiki/view/chan_dahdi.conf but it looks old |
| 19:10.59 | wastonNope | 2008 |
| 19:13.02 | ChannelZ | JoshBon: Loss is loss, packets that don't successfully make it from the source to the destination. I'm not sure what mtr is actually doing/sending |
| 19:14.11 | WIMPy | ICMP echo requests |
| 19:14.47 | ChannelZ | so I guess you could get false positives if someone doesn't respond to pings |
| 19:15.01 | ChannelZ | but the fact that your calls sounds like shit seems to indicate actual loss |
| 19:15.13 | ChannelZ | Of many possible problems I suppose |
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| 19:15.21 | JoshBon | Yeah |
| 19:15.31 | JoshBon | But I wondered what the loss is exactly |
| 19:15.42 | JoshBon | As it reports they also all go through :/ |
| 19:18.14 | WIMPy | That's just someone not playing along. |
| 19:18.41 | ChannelZ | I don't know what you mean by it shows they "all go through", what options are you using? |
| 19:19.09 | ChannelZ | All I see mtr report is the number of packets sent, a loss %, and latency. |
| 19:20.16 | WIMPy | You can have a loss to some node on the way without having loss to the destinaton. |
| 19:20.42 | JoshBon | I am running mtr -tu sip.server |
| 19:20.49 | JoshBon | They all get to the destination |
| 19:20.57 | JoshBon | But on one of the hops, there is a 90% loss |
| 19:21.03 | JoshBon | Loss of what, they all got there?! |
| 19:21.50 | WIMPy | That node doesn't reply to your requests. |
| 19:21.57 | WIMPy | Or rarely. |
| 19:22.14 | wastonNope | JoshBon: some routers depreference certain types of traffic and will drop them first if they are above a certain threshold of CPU for instance |
| 19:22.42 | wastonNope | JoshBon: it could be that router is dropping the specific type of packets that MTR is using for the probe |
| 19:23.24 | wastonNope | JoshBon: if you are able to adjust the type of probes in MTR (it's been a while) you should try that |
| 19:25.06 | JoshBon | Okay, I'll take a look. Currently looking at moving to a different server |
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