| 00:12.54 | SpaceInvaders | Does anyone know where I can find submenuopts? It's missing from the demo which I uncommented in the extensions.conf |
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| 02:00.42 | philfry | i'm having trouble with dahdi 1/1 but it seems dahdi 2/1 works just fine. Is there a way to force asterisk to use a certain analog line? I wanted to test the other phone lines remotely with originate in asterisk cli. |
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| 04:54.02 | FarhaadN | my asterisk proccess memory usage its problem |
| 04:54.08 | FarhaadN | top command shows me |
| 04:54.10 | FarhaadN | <PROTECTED> |
| 04:54.10 | FarhaadN | 31806 root 20 0 9328m 4.6g 10m S 35 39.6 92:14.04 /usr/sbin/asterisk -G asterisk |
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| 06:15.06 | ntz | hello |
| 06:15.56 | ntz | can I specify a subnumber for Dial() o.O ?? |
| 06:17.05 | ntz | i have faxes over t38/sip and faxmail application does something weird with the number and sends to asterisk 12345#679 (including `#' character) and asterisk can't parse that according to dialplan correctly |
| 06:18.07 | ntz | other app from the same suite does it somehow correctly and I can see on target device that the call was for 12345 with extension (subnumber or subaddress - terminology) `679' |
| 06:18.45 | ntz | can I somehow from AEL do a dial with subnumber - eg that I dis-assembly number with # character and call it properly ? |
| 06:20.34 | ntz | there's probably bug in the application because this form: ââuser@phone-number#subaddressââ |
| 06:20.42 | ntz | doesn't work from one but works from other |
| 06:21.11 | snadge | is /var/log/asterisk/master.db only used for sqlite cdrs? |
| 06:21.27 | snadge | im trying to understand why it exists.. and its open.. if the use count for the cdr sqlite3 module is 0 |
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| 11:59.56 | Haris | guys, has anyone here deployed an asterisk+freepbx setup for webrtc / video calls ? need help |
| 12:00.12 | Haris | or configured asterisk for webrtc/video calls |
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| 12:23.58 | doolittlework | hello |
| 12:25.04 | doolittlework | does anyone have any experience using a ata to receive a fax as tiff and store fax on ata for retrieval? |
| 12:25.13 | doolittlework | using https? |
| 12:35.14 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
| 12:35.37 | [TK]D-Fender | doolittlework, ATA's are not fax machines. |
| 12:40.56 | doolittlework | [TK]D-Fender: I am just trying to figure out how they get the faxing over https to work, they are using a ata. |
| 12:41.23 | [TK]D-Fender | Who? |
| 12:41.47 | doolittlework | http://www.faxsipit.com/ |
| 12:43.31 | [TK]D-Fender | Where does it say the ATA stores anything? |
| 12:44.26 | doolittlework | the only port that they require open is 443 and port 80, do i don't think they are using sip |
| 12:45.18 | [TK]D-Fender | maybe their device is doing an SSL VPN <- |
| 12:46.57 | doolittlework | i am sure it stored it somewhere, i have tested when i send a fax the far fax machine does not ring only rings and receives fax like 10 minutes after I have sent it |
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| 12:55.34 | [TK]D-Fender | So? that could be stored up ANYWHERE |
| 12:55.43 | [TK]D-Fender | like at the provider themselves queueing it up |
| 12:55.57 | [TK]D-Fender | no dumb ATA is a fax storage machine |
| 12:58.45 | Haris | what do I need to add to asterisk to make webrtc calls ? or make it easy for me to configure asterisk to make webrtc / browser to browser video calls ? |
| 12:59.27 | [TK]D-Fender | https://www.google.ca/#q=asterisk+webrtc+tutorial |
| 12:59.42 | [TK]D-Fender | Google is able to provide links to DOZENS of guides. |
| 13:00.06 | [TK]D-Fender | Including those on the official WIKI |
| 13:00.13 | [TK]D-Fender | Read there before asking here. |
| 13:01.07 | Haris | nothing that is practical. all theory |
| 13:01.19 | Haris | I'v read many of those articles |
| 13:02.05 | [TK]D-Fender | nothing practical? |
| 13:02.11 | [TK]D-Fender | The WIKI one shows a full setup |
| 13:02.15 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
| 13:02.16 | Haris | I need server side implementation THAT WORKS |
| 13:02.29 | [TK]D-Fender | including screenshots |
| 13:02.30 | Haris | not the client side part |
| 13:02.32 | Haris | hmm |
| 13:03.02 | [TK]D-Fender | I see extremely obvious client-side screenshots |
| 13:03.06 | [TK]D-Fender | You aren't looking |
| 13:03.13 | Haris | I am openign it |
| 13:03.14 | [TK]D-Fender | and that's just the FIRST link I checked on. |
| 13:04.03 | Haris | I think this is one of the first articles I read many weeks ago. going through it again |
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| 13:06.11 | Haris | I tried this with my manual installs of asterisk/freepbx. this doesn't work |
| 13:06.20 | [TK]D-Fender | Show us |
| 13:06.28 | Haris | I'v since moved on to distro install |
| 13:07.07 | Haris | I'v already shown results with my manual installs in past weeks |
| 13:07.27 | Haris | in this article, is it recommending to use pjsip for webrtc calls or chan sip ? |
| 13:07.28 | [TK]D-Fender | Don't waste our time if you have nothing to show |
| 13:07.46 | [TK]D-Fender | saying you showed "something" weeks ago is a waste of our time |
| 13:07.51 | Haris | its showing both |
| 13:08.05 | [TK]D-Fender | You also spent weeks just trying to follow instrtucitons to get it all installed |
| 13:08.17 | [TK]D-Fender | WEEKS |
| 13:08.32 | [TK]D-Fender | Don't tell us "doesn't work". |
| 13:08.38 | [TK]D-Fender | Show us actual proof |
| 13:08.57 | [TK]D-Fender | that means firewall dumps and everything |
| 13:09.13 | Haris | I installed distro. configured 2 extensions. enabled webrtc. recently installed ssl certs. still browser based clients can't register when I login via https to ucp |
| 13:09.30 | [TK]D-Fender | that is a useless garbage description |
| 13:09.31 | [TK]D-Fender | DEBUG. |
| 13:09.33 | [TK]D-Fender | CONFIGS |
| 13:09.38 | Haris | without https, ucp logins show registration attempt, but browser based clients go unreachable soon after |
| 13:10.04 | Haris | on asterisk cli |
| 13:10.07 | Haris | in debug mode |
| 13:10.16 | Haris | in debug mode = with sip debug on |
| 13:10.58 | ntz | [TK]D-Fender: servus .... JFYI, we're partially resolved our issue, this what we need: https://issues.asterisk.org/jira/browse/PRI-64 |
| 13:11.42 | ntz | [TK]D-Fender: just to remind you: our problem with asterisk supporting numbers in format [$who]@$base_num#$subaddress |
| 13:11.51 | [TK]D-Fender | ntz, Not sure I assisted with this issue previously... |
| 13:11.53 | Haris | what config should one show ? I can show screenshots of two extensions configured. webrtc enabled. ssl cert installed. sip config done. and still not working. what else ? |
| 13:12.06 | ntz | [TK]D-Fender: 3 days ago, you've helped me pretty much |
| 13:12.58 | [TK]D-Fender | Haris, see that guide? ALL THOSE PARTS OBVIOUSLY. Show us everything required for this setup to work. |
| 13:13.26 | Haris | that guide is bull crap. the software doesn't work as depicted in that article |
| 13:13.30 | Haris | I'm going to show my setup |
| 13:14.13 | [TK]D-Fender | "the software"? What "software"? |
| 13:14.51 | [TK]D-Fender | And that is the official WIKI written & maintained by Digium themselves. |
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| 13:15.06 | [TK]D-Fender | * Created by Rusty Newton, last modified on Oct 30, 2014 |
| 13:16.00 | Haris | freepbx config of asterisk in freepbx distro install |
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| 13:17.07 | [TK]D-Fender | You're in #asterisk now and we don't care at all about FreePBX here. |
| 13:17.20 | [TK]D-Fender | If it is generating configs wrong that's their problem, not ours |
| 13:17.36 | [TK]D-Fender | Show us that it ISN'T doing them wrong and tthings still don't work. |
| 13:18.11 | Haris | what config does it take to make webrtc calls work through asterisk ? I'll show that specific config |
| 13:18.15 | Haris | I'll omit the rest |
| 13:18.21 | [TK]D-Fender | READ THE GUIDE |
| 13:18.30 | [TK]D-Fender | it showed all of them |
| 13:19.38 | Haris | is bindaddr=:: ok in http.conf ? |
| 13:19.49 | Haris | or does it have to be bindaddr=0.0.0.0 |
| 13:20.14 | Haris | netstat shows ===> tcp 0 0 :::8088 :::* LISTEN 1888/asterisk |
| 13:20.35 | Haris | that part from http.conf ok to make webrtc work .. as per that guide ? |
| 13:20.53 | [TK]D-Fender | you just shoed that it DOESN'T match. |
| 13:20.57 | [TK]D-Fender | What do YOU think? |
| 13:21.34 | Haris | I think it should work, since :: is used by the OS for both IPv4 and IPv6 based connection(s) |
| 13:21.36 | [TK]D-Fender | Does that look like IPv4 to you? are YOU using IPv4 for this? |
| 13:21.41 | Haris | supported+ by default |
| 13:21.53 | [TK]D-Fender | "by default" is LAZY |
| 13:22.03 | [TK]D-Fender | and you aren't starting by actually following what it SAYS should work |
| 13:22.18 | Haris | ok. let me make that change |
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| 13:28.52 | Haris | made the change. also asked centos community. in so many words .. they say it doesn't make a difference. :: is primarily for IPv6, but has fallback for IPv4. so its essentially the same. if asterisk is not treating it as such, then its practically in error |
| 13:29.52 | Haris | let's see if calls work now, that bindaddr=0.0.0.0 |
| 13:29.57 | [TK]D-Fender | so far you showed 2 tiny little lines |
| 13:30.37 | [TK]D-Fender | and that took 20 minutes |
| 13:32.39 | Haris | my http.conf is as per that guide, with 3-4 extra options set. I don't have icesupport. does that matter for this one ? |
| 13:32.44 | Haris | i.e., webrtc calls |
| 13:32.53 | Haris | i.e., browser to browser video calls |
| 13:33.08 | Haris | if its an optional feature I can do without, I don't care for it |
| 13:33.53 | Haris | my http.conf and rtp.conf match the guide. checking on sip.conf |
| 13:33.59 | [TK]D-Fender | Show all of it or stop wasting our time. |
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| 14:13.47 | chuckf | pretty much TRex |
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| 14:22.06 | mub | GRRRRRR I hate SIP. Why wouldn't DIGIUM phones support IAX2?! |
| 14:22.08 | mub | They're DIGIUM |
| 14:22.19 | mub | And they're supposed to be best buds with asterisk |
| 14:29.27 | [TK]D-Fender | IAX is not designed for phones |
| 14:29.33 | [TK]D-Fender | They even hid that fact in the name |
| 14:32.02 | WIMPy | Yes, that's absolutely hilarious. |
| 14:32.43 | WIMPy | IAX might not have been designed for phones, but it's surely much better at it than sip. |
| 14:33.05 | WIMPy | But I have to admit that it lacks status subscription support. |
| 14:34.28 | mub | It's MUCH better. I have to configure weird NAT settings on all of these phones just to make it work with my asterisk box at a different location... IAX2 softphones though? they just work. |
| 14:34.29 | WIMPy | And I'm not even sure about the not designed for phones part. Back in the days there was that IAXy thing, which is kind of a phone. |
| 14:35.08 | WIMPy | It's not just NAT, although that's already a real killer argument on its own. |
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| 17:26.40 | rrittgarn | when using app_queue, trying to specify a rule that increases minimum penalty, rule show s up in queue show rules. Relevant configs: http://pastebin.com/5zmCdtjF |
| 17:27.13 | rrittgarn | so clarification / problem i need is - the first call that the rule is adjusted on - does that get affected by the rule change? or no, only subsequent callers? |
| 17:27.24 | rrittgarn | because it looks like its only subsequent callers or I'm missing / screwed something up |
| 17:27.40 | rrittgarn | (asterisk version 13.5) |
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| 18:04.45 | rrittgarn | after further reading/googling i see that queuerules don't behave the way i thought - only new calls are affected - is there a better solution than just dumping into another queue with more members ? Looking at keeping some of the reporting straightforward if possible |
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| 18:54.28 | hdon | hi all :) if # is my feature code for transfer, is there another way i can send the dtmf #? |
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| 19:17.12 | [TK]D-Fender | What kind of phones are you using? |
| 19:27.07 | Kalavera | [TK]D-Fender: Hello , here is the call that has audio only one way : http://pastebin.com/B7UHrW4s |
| 19:29.15 | [TK]D-Fender | You are only looking at HALF of the call |
| 19:30.39 | Kalavera | [TK]D-Fender: there is a lot of traffic and lots of extensions , I have done that using sip debug set peer 1596 |
| 19:31.09 | Kalavera | [TK]D-Fender: but there was some other lines related to other calls that I have deleted |
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| 19:39.46 | Kalavera | [TK]D-Fender: here is a tcpdump from the beginning to end of call : https://bpaste.net/show/05cb8c24072c |
| 19:40.16 | Kalavera | :q |
| 19:41.13 | [TK]D-Fender | I don't want external dumps... |
| 19:41.19 | [TK]D-Fender | * clive SIP w/ SIP debug |
| 19:41.24 | [TK]D-Fender | complete call, beginning to end |
| 19:47.50 | *** join/#asterisk CeBe1 (~CeBe@a81-14-235-90.net-htp.de) |
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| 19:49.52 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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| 20:18.18 | mxmasster | hello |
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| 20:52.38 | Kalavera | hey guys I am having a one way audio problem here is the call track https://bpaste.net/show/58a74c0e0d0f , any sufestions / help will be good |
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| 21:10.42 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
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| 21:21.56 | Kalavera | [TK]D-Fender: https://bpaste.net/show/58a74c0e0d0f |
| 21:23.06 | [TK]D-Fender | [TK]D-Fender |
| 21:23.06 | [TK]D-Fender | I don't want external dumps... |
| 21:23.06 | [TK]D-Fender | 3:41 pm |
| 21:23.06 | [TK]D-Fender | * clive SIP w/ SIP debug |
| 21:23.06 | [TK]D-Fender | 3:41 pm |
| 21:23.07 | [TK]D-Fender | complete call, beginning to end |
| 21:24.31 | mxmasster | I was looking into g729 is the patent expiring this month? |
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| 21:35.15 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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