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| 00:54.28 | jayvee | 14:19 < Penguin> Raise your concern during regular US business hours. |
| 00:54.32 | jayvee | Penguin: were you being sarcastic? |
| 00:54.57 | jayvee | 14:20 < [TK]D-Fender> jayvee: Show us an actual problem now. |
| 00:55.33 | jayvee | [TK]D-Fender: I'm more interested in having a response on the bug report itself. I'm not going to respond to it until it's reopened. Like I said, what's the best way to get it reopened? Or should I file a new ticket? |
| 00:56.10 | jayvee | If posting on the bug report was going to work, then there would have been a post on it by now saying something along the lines of what you've said. |
| 01:18.14 | jayvee | Just to clarify, I'm more than happy to do what you've said, but I need some kind of assurance I won't be wasting my time |
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| 01:29.37 | [TK]D-Fender | jayvee: There is no re-opening on an expired bug report like that because the very first thing you'll have to to do is upgrade and prove a case for the current version |
| 01:30.19 | [TK]D-Fender | jayvee: nobody is fixing a version from 2 years ago which was very poorly presented in the first place. |
| 01:31.54 | [TK]D-Fender | jayvee: So go update and see if things don't appear to be working as expected and do a proper job of backing up the claim. |
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| 07:07.14 | nunne | anyone have had any chance yet to try pjsip in asterisk 12/13? does it do a better job at handling more registrations (making having a sip proxy perhaps obselete in *that* regard) |
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| 07:39.20 | grzew | hello @all |
| 07:39.38 | grzew | I have a question about dtmf. |
| 07:40.12 | grzew | When I place call from console: console dial number I can see in console that asterisk is receiving dtmf codes. |
| 07:40.58 | grzew | The problem is when I place call: channel originate SIP/trunkname/number application Macro macroname,args |
| 07:41.23 | grzew | I canât see any received dtmf codes |
| 07:41.54 | grzew | Also this call is not stored in cdr records. |
| 07:42.09 | grzew | Can someone help me with dtmf issue? |
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| 09:00.38 | grzew | Maybe someone can help me here: http://forums.digium.com/viewtopic.php?f=1&t=91235&sid=38073b3145d2f3e81e22bcd0ef756974 |
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| 09:42.11 | lantizia | Anyone know which variety of a snom 320 is in this video? http://www.youtube.com/watch?v=nUfDN59PTvc |
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| 10:27.57 | rhineheart_m | may I know how to do debugging? Like I want to know why my calls to pstn is being dropped |
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| 10:41.39 | rhineheart_m | what does it mean with Sending fake auth rejection for device? thanks |
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| 12:41.25 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.12.0 (2014/08/19), 1.8.30.0 (2014/08/19); Standard: Asterisk 12.5.0 (2014/08/19); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 13:41.49 | marceloamorim | guys, I had an issue with my audio when I call from my asterisk to someone, I use asterisk 11.6-cert4 on debian 7.4 |
| 13:43.09 | marceloamorim | I don`t hear anything, and there isn`t nat on the network |
| 13:44.28 | marceloamorim | the [tk] look into the debug and looks fine, so he think about digium device or something like that, but I just test and the hardware its fine |
| 13:45.48 | marceloamorim | I use ip-tunnel on the client and the server, do you know what might be? |
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| 14:14.38 | LiohAu | is the IAX protocol an equivalent of the SIP protocol? and if yes, is it based on RTP also? |
| 14:15.19 | WIMPy | No, it's much more well thought out and no, there's no RTP. It handles media as well. |
| 14:18.10 | LiohAu | WIMPy: iâm trying to make a remote-controlled drone and I would like to transmit image and voice, what protocol would you suggest ? SIP, IAX or XMPP (jingle?) |
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| 14:29.25 | marceloamorim | WIMPy: I think I found my problem, but I can`t test right now |
| 14:30.52 | marceloamorim | directmedia should set outgoing, because I routed the blocks and there is no nat on my network |
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| 14:56.30 | WIMPy | LiohAu: Why do you want to make it that complicated? |
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| 15:13.00 | LiohAu | WIMPy: what do you mean by `that complicated` ? |
| 15:13.50 | LiohAu | It seems easier to use an existing protocol instead of sending images/sound my self with a basic UDP socket no? |
| 15:15.00 | WIMPy | What about a normal tcp stream? |
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| 15:35.00 | LiohAu | WIMPy: latency |
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| 15:35.58 | LiohAu | moreover I guess that the protocols are handling lot of things that I would not handle, my goal is not to reinvent the whole wheel |
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| 16:10.32 | WIMPy | well, if you need low latency realtime, RTP or IAX would be an option, yes. |
| 16:12.03 | WIMPy | feels wierd about having said that. |
| 16:19.36 | LiohAu | WIMPy: can you tell me more about security aspects ? I saw that I can use ZRTP instead of RTP, but are there equivalent solutions for IAX ? |
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| 16:20.54 | WIMPy | -Asterisk can't do ZRTP, but SRTP should be good enough for you. And IAX has encryption as well. |
| 16:21.53 | WIMPy | And you could always use openvpn or fastd or something that would allow you to change connection without interrupting the stream. |
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| 16:31.08 | LiohAu | WIMPy: from what I read the difference between ZRTP and SRTP is not really huge (a hash is sent + compared for each packet to prevent its modification by an attaker) |
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| 16:33.28 | LiohAu | btw can you highlight me when you answer, so maybe weâll be able to chat in real time :D |
| 16:33.40 | LiohAu | youâll* |
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| 16:35.46 | WIMPy | LiohAu: the difference is that ZRTP is end-to-end and SRT is only for one link. But if you have only one link... |
| 16:37.41 | LiohAu | WIMPy: So if iâm doing my stuff in a personal LAN SRTP is enough, while if Iâm using internet ZRTP will keep everything secure on the multiple relay used ? |
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| 17:49.07 | WIMPy | LiohAu: It does't matter what the IP network looks like. It's about SIP proxies. |
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| 19:43.45 | malachi_constant | Hi guys. |
| 19:44.41 | malachi_constant | I'm doing an installation using a rhino channel bank and I'm trying to see if I've terminated everything correctly on the feed side. |
| 19:45.20 | malachi_constant | It goes 66 block --> Rhino CB24-FXO --> Sangoma T1 card. |
| 19:46.48 | malachi_constant | When I call it rings the line but is there any way for me to see that it's hitting the Rhino? |
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| 19:55.38 | malachi_constant | My apologies if anyone replied, I had to reboot this machine. |
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| 20:34.22 | tinklebear | I totally got googletts working. It's so cool. Programatically making calls with node.js. Too cool. |
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| 20:40.27 | pabelanger | tinklebear, good to see it still working. Actually going to something with it too in the next week or so |
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| 20:49.36 | sklav | hi guys having a weird issue |
| 20:50.15 | sklav | when calling my DID from outside the network all i hear is silence but the phone actually is ringing and if i answer all is good |
| 20:50.22 | sklav | what can cause this issues? |
| 20:52.57 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 20:55.51 | ChannelZ | Depends. Sounds like maybe your ITSP isn't generating the ringing on call progress |
| 20:57.20 | ChannelZ | Or, in your dialplan, are you Answer()ing prior to Dial()ing the device? |
| 20:58.47 | sklav | seems im answering before i dial |
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| 20:59.34 | sklav | actually i just added the following n,Dial(SIP/12602,30,mKkTt) and i can hear my musiconhold |
| 20:59.37 | sklav | weird |
| 20:59.51 | sklav | im not hearing the ringtone which would be expected |
| 21:00.40 | ChannelZ | If you're not playing any prompts or anything, don't answer first. |
| 21:00.57 | sklav | how do i change that? |
| 21:01.30 | sklav | i see ok i think i figured it out |
| 21:03.22 | sklav | figured it out |
| 21:03.33 | sklav | thanks for the right pointer |
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| 22:19.09 | malachi_constant | Okay so, channel bank seems to be fine, but when I try to call out from the CLI I get 'unable to request channel DAHDI/1' |
| 22:20.11 | malachi_constant | Is that an issue internal to asterisk/dahdi/libpri? |
| 22:20.16 | malachi_constant | Or is it something upstream? |
| 22:22.21 | WIMPy | Are the channels configured correctly? What does 'dahdi show ...' tell you? |
| 22:25.07 | malachi_constant | WIMPy: dahdi show channels shows 24 channels and the pseudo-thing. |
| 22:25.23 | malachi_constant | dahdi show status shows the wanpipe driver for my sangoma card. |
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| 22:26.59 | malachi_constant | WIMPy: 'originate DAHDI/1 extension myphonenumber' is the command I'm using. |
| 22:28.11 | WIMPy | Looks ok so far. Anything showing up when you crank up verbose and debug? |
| 22:28.19 | malachi_constant | WIMPy: I will check it out. |
| 22:34.31 | malachi_constant | WIMPy: I set both verbose and debug to 9 in asterisk.conf. |
| 22:35.00 | WIMPy | Next time you can use 'core set ...'. |
| 22:35.15 | malachi_constant | Ooooh. Right! |
| 22:35.20 | malachi_constant | Fortunately no users on this system yet. :-) |
| 22:36.20 | malachi_constant | WIMPy: Same stuff in /var/log/asterisk/messages. Is there somewhere else I can find more information? |
| 22:36.48 | WIMPy | Do you log debug at all? Watch on the CLI while you test. |
| 22:52.52 | malachi_constant | WIMPy: with full => notice,warning,error,debug,verbose,dtmf,fax that's still all I get. |
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| 22:53.53 | malachi_constant | oooooh |
| 22:54.16 | malachi_constant | 'Channel 1 off hook, can't use' |
| 22:56.07 | malachi_constant | But it goes to a Rhino channel bank, which is connected via RJ-21/amphenol to a 66 block. |
| 22:56.31 | malachi_constant | FXO channel bank specifically. |
| 22:57.52 | malachi_constant | Maybe it's just the signalling setting? I'm using FXO ground start. |
| 23:02.53 | malachi_constant | 'Device DAHDI/24 changed to state '2' (In use) but we don't care because they're not a member of any queue." |
| 23:04.37 | malachi_constant | that's following me changing the channel bank to 'immediate'. |
| 23:06.03 | malachi_constant | Do channel banks like this "translate" signalling to ground start/loop start/whatever, or does it expect it to already be in that format? And if so, should I configure the channel bank to have 'Protocol: Ground Start' on the T1 interface? |
| 23:07.23 | malachi_constant | I think the telco is using ground start. |
| 23:07.57 | WIMPy | I have absolutely no idea. |
| 23:07.58 | malachi_constant | (Sorry for the volume of questions. I'm googling frantically and trying to approach understanding.) |
| 23:08.09 | malachi_constant | It would be better if it was 100% SIP. |
| 23:08.27 | WIMPy | That gives other headaches. |
| 23:09.16 | WIMPy | But the voltage/current detection must happen inside the channelbank. So I guess that has some basic configuration, |
| 23:09.18 | WIMPy | . |
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