| 00:11.42 | *** join/#asterisk Katty (uid62315@gateway/web/irccloud.com/x-aatntlreqzwzlqcy) |
| 00:12.38 | *** join/#asterisk _root_ (~slmn@unaffiliated/root/x-2442832) |
| 00:12.44 | _root_ | hello |
| 00:13.17 | _root_ | I though for a bit that I need freepbx or other gui to work with asterisk |
| 00:13.47 | _root_ | Could some one help me for material on how to work with asterisk in terminal |
| 00:13.51 | _root_ | thanks |
| 00:19.21 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:19.21 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 00:31.08 | *** join/#asterisk matrix1233 (~matrix123@41.230.40.216) |
| 00:34.18 | *** join/#asterisk [[thufir]] (~thufir@192.157.119.2) |
| 00:49.59 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 00:56.27 | *** join/#asterisk gnomethrower (~Wings@unaffiliated/gnomethrower) |
| 00:56.33 | gnomethrower | Hey guys |
| 00:56.39 | gnomethrower | I have a Polycom phone I'm using with Asterisk |
| 00:56.50 | gnomethrower | we have 3-digit extensions, eg, 100, 110, 111 |
| 00:57.11 | gnomethrower | when I try to transfer to anything 10* or 11* it erroneously tries to dial |
| 00:57.14 | gnomethrower | but 12* is fine |
| 00:57.24 | gnomethrower | my dial-plan is such: |
| 00:57.24 | gnomethrower | [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT|**x.T |
| 01:07.45 | Samot | Do you mean 100 or do you actually mean 10* |
| 01:07.58 | Samot | Is the * the actual asterisk or are you just using that as a wildcard? |
| 01:10.44 | gnomethrower | Samot: Using it as a wildcard |
| 01:10.46 | gnomethrower | sorry to be unclear |
| 01:10.53 | gnomethrower | like, extension 101, 100, 110, 120 |
| 01:11.01 | Samot | asterisk -rvvvvvvvvvvv |
| 01:11.06 | gnomethrower | I think I found a "solution" |
| 01:11.06 | Samot | ~pb |
| 01:11.06 | infobot | i guess pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 01:11.14 | Samot | show a call this is happening on |
| 01:11.28 | gnomethrower | Samot: I ended up just disabling digit map on the Polycom phones |
| 01:11.31 | gnomethrower | and letting Asterisk handle it |
| 01:11.36 | gnomethrower | which seems to work perfectly |
| 01:12.01 | gnomethrower | turns out most phones in this office already had digit map turned off and those work fine |
| 01:12.18 | Samot | OK |
| 01:12.29 | gnomethrower | Thanks anyways! :) |
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| 01:31.21 | *** mode/#asterisk [+o gtjoseph] by ChanServ |
| 01:51.49 | *** join/#asterisk dunderproto (~dunderpro@122-117-44-11.HINET-IP.hinet.net) |
| 02:09.24 | dunderproto | Is it on-topic to ask help for general SIP related telephony problems in this channel? |
| 02:10.09 | Samot | Just ask |
| 02:11.36 | dunderproto | Samot: Thanks. I was trying to set up my 2 sip clients to call each other, one on a phone and another on my laptop. Both are behind NATs so I used a stun server. I was delighted to find that it worked when the phone called the laptop (the audio was great), but not the other way around (only silence) |
| 02:12.26 | dunderproto | when the phone calls the laptop, *both* sides have good audio. But when the laptop calls the phone, *neither* side has audio |
| 02:12.30 | Samot | That is a NAT issue |
| 02:12.48 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 02:13.13 | Samot | Look at the firewall/router for the one that has inbound issues |
| 02:13.59 | Samot | A STUN server is going to clean up the SIP packet |
| 02:14.15 | Samot | And the routing once it leaves the network.. |
| 02:14.36 | Samot | Still has to send in for new calls and thats firewall/nat |
| 02:20.17 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 02:21.32 | dunderproto | Samot: ok, thanks, I'm going to experiment with it |
| 02:31.34 | *** join/#asterisk matrix1233 (~matrix123@41.230.40.216) |
| 02:32.19 | dunderproto | Samot: the one with issues allows all outgoing traffic (and associated replies) but blocks all new incoming traffic |
| 02:32.39 | Samot | Yes, that would be a firewall/NAT issue. |
| 02:33.03 | Samot | When I say NAT I mean the actual process of the network address translation |
| 02:33.10 | dunderproto | What ports would I need to open up for sip to work? |
| 02:33.23 | Samot | Either there isn't a NAT rule to handle it or the firewall isn't allowing it |
| 02:33.29 | Samot | I don't know. |
| 02:33.35 | Samot | I don't know your setup. |
| 02:33.49 | Samot | I don't know what is what and what you have them set to. |
| 02:34.24 | dunderproto | Samot: I see. OK, I'll do some research |
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| 03:32.46 | dunderproto | Samot: OK, I managed to get it to work when I disabled NAT and just plugged the phone straight into the internet |
| 03:32.50 | dunderproto | two way calling, hurray |
| 03:32.57 | dunderproto | but I really would like the firewall back :) |
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| 03:35.25 | Samot | Uhm. |
| 03:35.27 | Samot | I wouldn't do that |
| 03:35.29 | Samot | At all. |
| 03:35.31 | drmessano | dunderproto: Buy a Mikrotik |
| 03:35.35 | Samot | Your phone is exposed. |
| 03:35.36 | Samot | Yes. |
| 03:35.41 | Samot | Buy a Mikrotik. |
| 03:35.44 | drmessano | routerboard.com |
| 03:36.23 | Samot | Phones/ATAs directly on the Internet are the source of fraud more than a hacked PBX. |
| 03:38.33 | dunderproto | Samot: Yeah, I definitely want a firewall |
| 03:38.44 | dunderproto | but I was happy to see it at least work. So it's clearly just a firewall issue |
| 03:38.44 | drmessano | routerboard.com |
| 03:38.54 | drmessano | ^^^^^^^^^^^^^^^^^^^^^ |
| 03:39.00 | drmessano | !!!ONES!!!111!!!!!! |
| 03:40.03 | Samot | dunderproto: Two endpoints doing direct media and both are behind NAT. All audio issues are basically firewall/NAT |
| 03:40.16 | Samot | There is a reason that two NAT endpoints don't do direct media. |
| 03:40.24 | Samot | It's a PITA to make work consistently. |
| 03:41.22 | Samot | Well or codecs. Firewall/NAT/Codecs. Those are going to be the source of audio issues in cases like this. |
| 03:42.12 | Samot | So the three causes of audio issue are Firewall, NAT and Codecs.... |
| 03:42.23 | Samot | And poor bandwidth... |
| 03:42.43 | Samot | So the FOUR causes of audio issues are Firewall, NAT, Codecs AND poor bandwidth... |
| 03:43.10 | Samot | Are you understanding? |
| 03:43.16 | Samot | Because if not.... |
| 03:43.20 | Samot | gets the comfy chair... |
| 03:43.28 | dunderproto | Samot: sorry, I went to get some food :) |
| 03:43.39 | Samot | All of that was a waste. |
| 03:43.43 | Samot | Thanks. |
| 03:43.44 | dunderproto | I've not had any problems with codecs/bandwidth |
| 03:43.57 | dunderproto | Samot: No, I'm definitely listening, my irc client has got history |
| 03:44.03 | dunderproto | just firewall/nat |
| 03:44.20 | Samot | Nobody expects just firewall/nat... |
| 03:44.44 | dunderproto | Samot: Sorry, please don't give up on me, I definitely listen to your input. Every bit of help is appreciated |
| 03:45.24 | Samot | Still nothing? |
| 03:45.25 | Samot | Wow. |
| 03:45.27 | Samot | OK. |
| 03:45.45 | Samot | should have gone with Holy Grail |
| 03:46.08 | dunderproto | ? I don't quite follow |
| 03:46.34 | Samot | I was paraphrasing The Spanish Inquisition from Monty Python |
| 03:47.48 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 03:49.46 | dunderproto | drmessano: thanks for the link |
| 03:50.17 | Samot | "Nobody expects the SPANISH INQUISITION!! Our MAIN weapon is fear and surprise! No, our TWO weapons are fear, surprise and ruthless efficiency....our *THREE* main weapons are fear, surprise, ruthless efficiency and an almost fanatical devotion to the pope......" |
| 03:52.43 | Samot | Oh Monty Python, I don't think there will ever be a chemistry like yours again. Some have come damn close but no.... |
| 03:54.56 | Samot | Of course being comprised of basically all first generation post-WWII Brits was a lot of the fuel. |
| 03:55.25 | Samot | And one American. |
| 03:55.49 | Samot | Who I have no doubt did A LOT of drugs. |
| 03:55.59 | Samot | Look at his animations.... |
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| 04:50.32 | keanne | hi, is asterisk capable of using gsm modems as gateway for outgoing calls? my gsm modem is wavecom multiband 900e 1800 as identified by gammu |
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| 05:16.14 | dunderproto | Samot: Thanks for your help. I got it to work by enabling ICE in addition to the stun server. Now both are behind firewalls and still connecting |
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| 08:03.53 | irc08153 | Asterisk 10.9.0, register => "number":"password"@sip.provider:5064/number~120, but asterisks keeps sending after reload/restart still to port 5060 (https://pastebin.ca/3836866) Is this a known bug? |
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| 08:29.46 | *** join/#asterisk DaRock (~Thunderbi@mail.unitedinsong.com.au) |
| 08:30.00 | DaRock | Hi guys |
| 08:30.30 | UncleKiwi | it doesnt matter what your name is |
| 08:30.34 | DaRock | Can someone please put me out of my misery and tell me how the hell to get rid of stdexten? |
| 08:31.40 | DaRock | I'm trying to dial using alphanumeric characters, so dial 'user', but for some damned reason it always defaults to some screwed up stdexten macro |
| 08:32.01 | DaRock | I've removed stdexten completely, and it still attempts it |
| 08:32.30 | DaRock | nothing on google is helping or clear on a resolution for that matter |
| 08:38.01 | *** join/#asterisk seik0 (3eb61f96@gateway/web/freenode/ip.62.182.31.150) |
| 08:40.11 | seik0 | Hi! I have asterisk 1.8 (I know what you think). Have success sip register to sip-provider, but upon register peer is not show in "sip show peers" and incoming calls a rejected with "No matching peer found". But if I do "sip show peer my_peer load" (to load this specific peer), then everything works. |
| 08:41.58 | seik0 | I found some issue: https://issues.asterisk.org/jira/browse/ASTERISK-12991, maybe this is the case. But, on the other hand, everything works on another asterisk with same version and same sip-provider and same sip-configuration (except, that problem asterisk is over nat, and ok asterisk is not). |
| 08:43.39 | seik0 | but in specified issue workaround is to user "type = friend" and it does not work for me |
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| 09:02.37 | irc08153 | Asterisk 10.9.0, register => "number":"password"@sip.provider:5064/number~120, but asterisks keeps sending after reload/restart still to port 5060 (https://pastebin.ca/3836866) Is this a known bug? |
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| 10:55.17 | irc08153 | Asterisk 10.9.0, register => "number":"password"@sip.provider:5064/number~120, but asterisks keeps sending after reload/restart still to port 5060 (https://pastebin.ca/3836866) Is this a known bug? |
| 10:58.33 | wdoekes | irc08153: you cannot reasonably expect us to look up the status of bugs in long-EOL non-LTS versions. try a recent non-EOL version and check if it's fixed there. if it's not, please file a bug report. |
| 11:04.11 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
| 11:08.28 | irc08153 | simply could have been that someone here have the knowledge and was also hit by this bug |
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| 11:52.54 | DaRock | so no idea why stdexten would be called without actually being in the dialplan? Or with other extension patterns to catch the dialled extension? |
| 11:53.41 | file | have you looked at "dialplan show" ? |
| 11:53.44 | file | it'll tell you where things come from |
| 11:55.21 | DaRock | I have... it simply doesn't exist in extensions.conf (at all), dialplan show doesn't bring it up, but if I dial an extension as 'user' it errors with some screwed up call to the stdexten macro |
| 11:55.55 | file | I'd suggest a pastebin of the console output and the dialplan show then... |
| 11:56.05 | DaRock | so I ran a dialplan show "user"@ and suddenly the call appears as priority 1! |
| 11:56.11 | DaRock | why? |
| 11:57.15 | DaRock | and I have tried patterns, even a straight exten => 'user' |
| 11:58.03 | DaRock | the only issue I can possibly see is if I adjust my priority for the pattern to 1 - but why is this even necessary? |
| 11:58.23 | Samot | DaRock: Let's see this dialplan. |
| 11:58.25 | Samot | ~pb |
| 11:58.25 | infobot | well, pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 11:58.27 | file | you're asking questions without giving me the needed information to be able to answer them |
| 11:59.56 | DaRock | well mainly the only thing to say is that this stdexten is a freaking poltergeist or something, keeps showing up unexpectedly |
| 12:00.52 | DaRock | actually... what part do I show here? dialplan show? or extensions.conf? |
| 12:00.56 | file | both |
| 12:00.58 | file | and console output |
| 12:01.21 | Samot | I want to see whats in extensions.conf that you wrote for this |
| 12:03.18 | DaRock | have any of you guys still have stdexten context in your dialplan? |
| 12:04.16 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
| 12:04.59 | DaRock | so console output: https://pastebin.com/AxGkWx85 |
| 12:05.43 | Samot | No. |
| 12:05.48 | Samot | That's not your dialplan. |
| 12:05.51 | Samot | That's the debug. |
| 12:06.02 | Samot | 8:01:24 AMÂ <Samot>Â I want to see whats in extensions.conf that you wrote for this |
| 12:06.12 | file | Samot: I asked for everything |
| 12:06.13 | DaRock | I know - its the console output asked for |
| 12:06.22 | DaRock | still working on dialplan |
| 12:06.32 | file | the console output shows what Asterisk did, dialplan show is how Asterisk interpreted the config, and extensions.conf is the underlying config |
| 12:06.35 | file | so you get a full view |
| 12:06.37 | DaRock | that was just the quickest |
| 12:09.08 | Samot | Well then..I'm going to the store. |
| 12:09.13 | Samot | No sense in waiting around.. |
| 12:15.49 | DaRock | dialplan: https://pastebin.com/wKjt6C6U |
| 12:16.23 | DaRock | sorry - lots of private stuff in there that can't go blowing about the net. Surely you understand that? |
| 12:17.42 | file | your configuration for the user extensions in the default context is calling the 'stdexten' subroutine. |
| 12:20.51 | file | you aren't using users.conf are you? |
| 12:21.42 | file | if so, it will automatically do things such as putting that into the created dialplan for dialing the user |
| 12:22.53 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:26.11 | DaRock | extensions.conf: https://pastebin.com/PzzhDBhQ |
| 12:26.32 | DaRock | where does it do or even say that? docs I mean |
| 12:27.09 | DaRock | and as such how do adjust that behaviour - not saying that its necessarily bad either |
| 12:27.09 | file | I have no idea, users.conf is over 10 years old and hasn't been used in most of those years |
| 12:27.19 | file | it's not maintained or recommended to be used |
| 12:27.49 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 12:27.51 | file | it is hardcoded in, it will either use that if "hasvoicemail" is set or otherwise it will add a priority that does a Dial() |
| 12:27.53 | DaRock | why? its still in docs, and how else do you maintain users across services like sip and iax? |
| 12:28.11 | file | you create them in the respective config files for each piece of functionality? |
| 12:28.45 | DaRock | so no central user system? Isn't that what users.conf was even for? |
| 12:29.01 | file | that was the goal of it but in practice it didn't work out |
| 12:29.31 | DaRock | so when did that memo come out? :-) |
| 12:29.40 | Samot | Years ago |
| 12:29.45 | Samot | Literally. |
| 12:29.50 | DaRock | no reference? |
| 12:30.11 | Samot | It's something that's been like this for a decade. |
| 12:30.14 | file | the code is still there and it works, but you are locked into how it does things and it won't be added to any subsequent stuff |
| 12:31.02 | DaRock | there's literally no indication of that in the docs |
| 12:31.38 | *** join/#asterisk matrix1233 (~matrix123@41.230.38.112) |
| 12:31.46 | file | it predates when we pushed hard on documentation |
| 12:31.57 | DaRock | that is really bizarre.... well at least now I know. Thanks |
| 12:32.52 | DaRock | really?? How are you supposed to learn this stuff? The docs even show user.conf. I hadn't even started this 10 years ago |
| 12:33.02 | Samot | DaRock: It works. |
| 12:33.05 | Samot | It can be used. |
| 12:33.16 | Samot | It's just not by the average person using Asterisk. |
| 12:33.31 | *** join/#asterisk nix8n82 (~AndChat58@67-130-74-235.dia.static.qwest.net) |
| 12:33.32 | Samot | It was a poor solution to a problem 10 years ago. |
| 12:33.36 | DaRock | but ti can apparently screw things up and leave you banging your head for days :-) |
| 12:33.41 | Samot | But some still like it and it works. |
| 12:33.49 | DaRock | it |
| 12:33.59 | Samot | Well it shouldn't be used for non-"expert" installs. |
| 12:34.10 | Samot | In the grand scheme it's more of a PITA then the normal way |
| 12:35.16 | Samot | Also, what version of Asterisk are you running? |
| 12:35.35 | DaRock | well thanks guys. At least I know where things are at now. Perhaps some notation in the docs might be more helpful ;-) |
| 12:35.54 | Samot | What notation? |
| 12:35.56 | DaRock | I'm running 11 atm - I have to update, I know |
| 12:37.20 | DaRock | While I'm at it, is there any good tuts on using asterisk with webrtc? I haven't seen much and in the docs it just points to third party libraries and apps |
| 12:37.36 | file | define "using asterisk with webrtc" |
| 12:37.51 | file | we have a wiki page for setting it up and using a web based SIP client as an example |
| 12:38.16 | file | WebRTC itself is just a tool though |
| 12:39.20 | *** join/#asterisk matrix1233 (~matrix123@41.230.38.112) |
| 12:39.46 | DaRock | How would I create my own webapp using webrtc in a browser to connect to asterisks webrtc service? |
| 12:40.30 | file | you'd presumably use a library and build a webapp around it |
| 12:41.06 | file | I'm not aware of any tutorials for that though, the information and the pieces are all out there though |
| 12:41.12 | DaRock | is the api standard? The docs asterisk don't go into too much detail other than using an app to use it - not what calls are required for operation |
| 12:41.40 | DaRock | and indications are in the docs that not all libs will work |
| 12:41.55 | file | Asterisk doesn't care, to it you are speaking SIP |
| 12:42.04 | file | the rest is on the web side |
| 12:42.11 | DaRock | thats what I thought might be happening |
| 12:42.44 | Samot | WebRTC is just a softphone in a browser. |
| 12:43.01 | file | Samot: that's what a lot of people make of it |
| 12:43.02 | Samot | Using web sockets for signaling. |
| 12:43.03 | DaRock | but I'm not 100% sure thats what the browser webrtc is expecting |
| 12:43.13 | Samot | WebRTC is a term |
| 12:43.20 | Samot | It's not really "a thing" |
| 12:43.30 | file | it's a Javascript API and then SDP |
| 12:43.34 | Samot | ^^^ |
| 12:43.34 | DaRock | actually its an api |
| 12:43.58 | file | the SDP we produce, when Asterisk is properly configured, works |
| 12:44.00 | Samot | WebRTC has some maturing to do. |
| 12:44.24 | Samot | Cuz there is no standard. |
| 12:44.32 | DaRock | it seems webrtc works well, but it might not be with asterisk |
| 12:44.43 | Samot | I use it with Asterisk, works fine. |
| 12:44.51 | Samot | Not that much, not a lot of call for it... |
| 12:44.53 | DaRock | did you make the app? |
| 12:44.57 | Samot | No |
| 12:45.10 | *** join/#asterisk nix8n82 (~AndChat58@67.130.74.235) |
| 12:45.21 | DaRock | well then... it might work with one app, and not another |
| 12:45.24 | Samot | But again, browsers are one of the biggest problems with WebRTC. |
| 12:45.31 | file | sighs |
| 12:45.32 | DaRock | possibly |
| 12:45.41 | file | the information that is exchanged is SDP |
| 12:45.42 | Samot | Chrome has broken quite a few WebRTC apps. |
| 12:45.50 | file | which the browser consumes and provides |
| 12:45.53 | DaRock | but clarity on what works with asterisk will certainly help |
| 12:45.54 | Samot | Yes, Asterisk handles the SDP |
| 12:46.00 | Samot | IT works. |
| 12:46.08 | file | the SDP we produce and accept works. |
| 12:46.16 | file | and in order for an app to use WebRTC, it has to do that |
| 12:46.22 | Samot | ^^^^^^ |
| 12:46.23 | DaRock | so just send the sdp? |
| 12:46.25 | file | it has to give us the SDP and it has to accept the SDP we provide and give it to the browser |
| 12:46.36 | Samot | And well signaling as well. |
| 12:46.47 | file | WebRTC has no defined signaling method, so some people wrote SIP stacks in Javascript to fill that part |
| 12:46.58 | file | Asterisk allows that by accepting it on the Websocket transport |
| 12:47.12 | DaRock | so why is webrtc considered separate to sip in asterisk then? |
| 12:47.26 | DaRock | scratch that |
| 12:47.27 | Samot | What do you mean? |
| 12:47.33 | Samot | Its not. |
| 12:47.43 | file | it's not separate, but it requires additional configuration because WebRTC mandates certain features |
| 12:47.55 | Samot | Like DTLS. |
| 12:47.59 | file | and ICE. |
| 12:48.01 | file | and RTCP-MUX. |
| 12:48.02 | DaRock | so thats the only diff - instead of 5060, it uses wss |
| 12:48.02 | Samot | Yup. |
| 12:48.10 | file | yes. |
| 12:48.21 | Samot | The port has nothing to do with the protocol..... |
| 12:48.43 | DaRock | yeah I know... jsut came to mind :-) |
| 12:48.46 | DaRock | just |
| 12:49.48 | DaRock | thats what's been throwing me - the wss/sip connection diff |
| 12:49.55 | Samot | No. |
| 12:49.58 | Samot | SIP is SIP |
| 12:50.07 | DaRock | would be useful to see some sample code |
| 12:50.10 | Samot | SIP can talk on UDP, TCP, TLS, WS, WSS |
| 12:50.16 | file | code for what exactly? |
| 12:50.22 | [TK]D-Fender | ~users.conf |
| 12:50.22 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 12:50.24 | file | the Javascript SIP clients provide examples |
| 12:50.25 | [TK]D-Fender | :) |
| 12:50.38 | file | https://tryit.jssip.net there's one for example |
| 12:51.19 | DaRock | for a webapp webrtc connection - not the media grabbing stuff, but just what would allow asterisk and browser to communicate effectively |
| 12:51.36 | DaRock | too tired to type apparently ;-) |
| 12:51.38 | Samot | Like any other client... |
| 12:51.50 | Samot | It sends requests to the host you define... |
| 12:52.21 | *** join/#asterisk nix8n82 (~AndChat58@67-130-74-235.dia.static.qwest.net) |
| 12:52.56 | file | there's nothing from an Asterisk perspective that would really be useful for us to provide, as we don't care provided it's SIP |
| 12:53.01 | Samot | Honestly, I haven't gotten a single shred of interest shown for WebRTC by anyone over the years. |
| 12:53.08 | Samot | As far as the average client goes. |
| 12:54.12 | Samot | WebRTC JS APIs have broken do to issues with browsers supporting the needed pieces for WebRTC |
| 12:54.27 | Samot | I believe Chrome had issues with the RTP-MUX stuff for a while.. |
| 12:55.04 | Samot | There's no "WebRTC" standard for browsers. |
| 12:55.12 | Samot | They aren't being designed with WebRTC in mind. |
| 12:55.25 | file | it's gotten better... |
| 12:55.35 | Samot | Yes. |
| 12:55.36 | DaRock | so intial connect to asterisk using wss://, usual sip signals, sdp, then a separate connection for rtp? That about it? |
| 12:55.36 | Samot | It has |
| 12:55.41 | Samot | Like I said, it needs to mature some more |
| 12:55.49 | file | they're still using SDP attributes which aren't in the specs last I checked |
| 12:55.54 | file | using and requiring, in some instances |
| 12:56.24 | Samot | WebRTC will be something if all the browsers got behind it and standardized for it. |
| 12:56.35 | Samot | But again, WebRTC isn't flooding the market really.. |
| 12:56.42 | DaRock | is it still under living standard or something? or is it now in the actual standard ecma? |
| 12:56.42 | file | DaRock: it's UDP so connectionless for RTP - but yes |
| 12:56.58 | DaRock | tls for the signaling then? |
| 12:57.05 | Samot | WS/WSS |
| 12:57.07 | DaRock | tcp I mean |
| 12:57.16 | Samot | Web Socket or Web Socket Secure |
| 12:57.33 | DaRock | yep got that :-D |
| 12:57.40 | *** join/#asterisk nix8n82 (~AndChat58@67-130-74-235.dia.static.qwest.net) |
| 12:57.49 | DaRock | like I said, too tired to type :-D |
| 12:58.53 | DaRock | Well thanks guys, I finally have something solid to work with |
| 12:59.16 | Samot | WebRTC is decent, but like I said..not a lot of demand for it. |
| 12:59.36 | DaRock | I know - too many scared away from it at this stage |
| 12:59.44 | file | it still gets used, just without people realizing it |
| 12:59.55 | Samot | Well yes. |
| 12:59.58 | file | it's not just audio/video, but also data |
| 13:00.00 | DaRock | I think I better get some sleep.... |
| 13:00.39 | DaRock | a lot of chats for web support, fb, google, etc... whiteboarding, screen sharing, vnc |
| 13:00.52 | DaRock | still hasn't peaked yet |
| 13:00.59 | Samot | Oh hey file.... |
| 13:01.05 | file | moo |
| 13:01.12 | Samot | I got an email from one of the freelancer sites.. |
| 13:01.40 | Samot | Some guy offered me $200 to rewrite the Asterisk Conference app to do video conferences and wallboard sharing... |
| 13:01.48 | Samot | TWO HUNDRED DOLLARS?!!?! |
| 13:01.48 | file | cute |
| 13:01.59 | file | we're already working on that |
| 13:02.04 | Samot | That's what I said. |
| 13:02.26 | Samot | Plus.. |
| 13:02.29 | file | the core and now PJSIP support multiple streams, as well as ConfBridge for SFU, working on the WebRTC side of things |
| 13:02.34 | Samot | TWO HUNDRED DOLLARS?!?!?!?! |
| 13:02.42 | file | yeah.... |
| 13:02.50 | Samot | That's like 4 hours of work. |
| 13:03.08 | file | I wish |
| 13:03.14 | Samot | No. |
| 13:03.16 | Samot | I mean for me |
| 13:03.21 | Samot | That's what $200 covers. |
| 13:03.22 | Samot | 4 hours. |
| 13:03.26 | file | oh |
| 13:03.28 | file | well, yes |
| 13:03.41 | Samot | Redoing the conf app and video stuff.. |
| 13:03.45 | Samot | Waaay more than 4 hours |
| 13:03.50 | Samot | I mean at least 6 right? |
| 13:03.52 | Samot | :) |
| 13:04.02 | file | adds stuff up |
| 13:04.11 | file | 2 people 3-4 months? |
| 13:04.20 | Samot | Oh yeah, I totally believe that |
| 13:04.44 | Samot | And that's if nothing other pulls at them.. |
| 13:04.59 | Samot | And there aren't other projects or issues that leak into their time. |
| 13:07.03 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 13:08.01 | file | disregarding the video side of it, the core work brings some nice possibilities to the mix |
| 13:08.14 | file | proper stream changing/adding/removing is a nice thing |
| 13:08.25 | Samot | Screenshare? Wallboard? |
| 13:08.29 | file | the foundation is there to actually allow the codec negotiation that people want |
| 13:09.00 | file | ie: remote side answers with a set of codecs, that could now easily get propagated to the caller and reflect in the answer to them |
| 13:09.08 | Samot | Well I'll keep you update on the new project for later this year. |
| 13:09.26 | file | or if someone turns on video we can reinvite to add the video stream to the other side |
| 13:09.39 | Samot | Have to do Property Management System integration and there's nothing for Asterisk. |
| 13:10.13 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 13:10.43 | Samot | Going to focus on the Mitel SX 2000 API and the Micros Fidelio API/protocol... |
| 13:12.16 | Samot | Luckily most of the stuff PMS had before as standard many don't use anymore. Call Rating, Mini-Bar charges, etc. |
| 13:12.25 | file | aye |
| 13:12.36 | Samot | They want to be able to "checkin/checkout" users, 911 alerts and wake up calls. |
| 13:12.53 | Samot | The checkin/checkout is where I'd have to go in and clear the VM, vm settings, etc, etc. |
| 13:13.07 | Samot | Dump any MWI.. |
| 13:13.44 | Samot | Anything that works with Asterisk right now requires middleware and some of it can get costly. |
| 13:14.19 | Samot | Since it's all Windows based. |
| 13:17.53 | *** join/#asterisk matrix1233 (~matrix123@41.230.38.112) |
| 13:18.40 | Samot | 9:09:03 AM F<@file> ie: remote side answers with a set of codecs, that could now easily get propagated to the caller and reflect in the answer to them <-- Wait... so I can ext 100 and send my codecs in the Offer and when they Answer I'll get to see what they answer with? |
| 13:18.49 | Samot | And choose the one I want to match with? |
| 13:19.29 | file | the foundation for it is now there |
| 13:19.33 | Samot | Nice. |
| 13:19.34 | file | it hasn't been done, but it's now not bad to do so |
| 13:19.40 | Samot | For the web stuff? |
| 13:19.51 | file | the multistream stuff isn't strictly for web |
| 13:20.21 | Samot | Well there would have to be a widget or something right? |
| 13:20.21 | file | the TLDR is that streams are now reflected internally, and events occur when they change/get accepted/etc |
| 13:20.31 | Samot | If I called from my phone... |
| 13:20.51 | Samot | How could I select the codec I wanted from the Answer? Would that be a widget or something that popups? |
| 13:20.52 | file | oh, I may have misunderstand you |
| 13:20.57 | Samot | OK. |
| 13:21.20 | Samot | I get that using a web conference or UI to make the call can give you codecs you want for audio/video... |
| 13:21.30 | Samot | I was just thinking it be nice for phones too. |
| 13:21.38 | file | Alice offer of ULAW/G722 into Asterisk, Asterisk calls device 'Bob' and offers ULAW/G722, Bob answers with 'G722' only, Asterisk could (with more effort), send back 'G722' only in the answer to Alice |
| 13:21.42 | file | right now it'd send back ULAW/G722 |
| 13:22.05 | Samot | Oh I see what you are saying. |
| 13:22.18 | file | once an answer occurs you'd have to do a reinvite to change, but that is also now possible |
| 13:22.21 | Samot | Asterisk can filter out the junk.. |
| 13:22.40 | Samot | That's still nice. |
| 13:22.47 | file | it's two different approaches - one is the "I just want stuff to work even if it means transcoding" and the other is "I want to avoid transcoding at all costs" |
| 13:22.58 | Samot | People never mess with their codecs and always sent 10,0000 at once |
| 13:23.12 | Samot | Out of order... |
| 13:23.14 | Samot | blah blah blah |
| 13:23.23 | Samot | This could help with that a lot. |
| 13:23.25 | file | Asterisk is presently "I just want stuff to work" which is why you encounter scenarios where it is transcoding when it could really probably reinvite and get away with not transcoding |
| 13:23.27 | *** join/#asterisk VTheThinker (4f726b43@gateway/web/freenode/ip.79.114.107.67) |
| 13:23.48 | file | the knowledge of that just wasn't in the core and accessible, and there was no interface to request such a thing |
| 13:24.01 | Samot | So basically, try not to transcode if possible but transcode if needed |
| 13:24.33 | file | yeah, the knowledge is now present to be more intelligent so it's possible ^_^ |
| 13:24.42 | Samot | That would be nice. |
| 13:24.46 | file | as I said though using the knowledge hasn't been done yet |
| 13:24.58 | Samot | Someone's gotta be the first. |
| 13:25.01 | Samot | Go for it |
| 13:33.14 | VTheThinker | I might be a little off-topic, but in case you know how I can solve this: I have an Openvox device http://www.openvox.cn/pub/manuals/old/English/VS-GW1600-20G%20User%20Manual.pdf with Asterisk installed; I performed a System Update and a System Reboot and now I cannot sign in anymore. I have the configuration backup since I downloaded it before upgrading... I tried some default credentials (admin/admin, admin/empty) but they do |
| 13:35.37 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 13:36.43 | *** join/#asterisk bford (d8cff501@gateway/web/freenode/ip.216.207.245.1) |
| 13:40.04 | Samot | VTheThinker: You're contacting them. |
| 13:40.11 | Samot | It's their appliance, their setup. |
| 13:49.32 | VTheThinker | the device is not at my office, it's at my client's offfice... and he';s not at the office right now. I guess he could solve it when he's there, right? |
| 13:49.45 | VTheThinker | but I will contact openvox anyway |
| 13:50.51 | Samot | We can't help with it. |
| 13:51.27 | VTheThinker | ok, thanks |
| 13:51.41 | Samot | Sorry, it's an appliance you can't log into.. |
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| 14:09.09 | VTheThinker | Samot do you mean it cannot be solved at all? It should have some default credentials... |
| 14:09.11 | *** join/#asterisk BoBeR182 (~BoBeR182@gateway/tor-sasl/bober182) |
| 14:09.14 | BoBeR182 | Hello! |
| 14:09.35 | Samot | No, I mean asking about it here isn't where you should be asking about it. |
| 14:09.37 | BoBeR182 | I was wondering what the minimum requirements for a running an asterisk server are |
| 14:10.07 | Samot | BoBer182: Depends on what it will be doing. How many calls, users, other things... |
| 14:10.38 | *** join/#asterisk kharwell (kharwell@nat/digium/x-gfqtiychwlyhopaz) |
| 14:10.38 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:10.40 | BoBeR182 | 1 user |
| 14:10.40 | VTheThinker | a, ok.. |
| 14:10.41 | *** join/#asterisk AndChat|589056 (~AndChat58@2600:100e:b003:3b2a:48b4:b9bd:f7e5:a642) |
| 14:10.43 | BoBeR182 | one call at a time |
| 14:11.08 | BoBeR182 | maybe up to 2 users during peak hours |
| 14:11.18 | Samot | A 1 core 1GB RAM will do it |
| 14:11.25 | BoBeR182 | VPS is fine? |
| 14:11.29 | Samot | yuo |
| 14:11.31 | Samot | yup |
| 14:11.45 | BoBeR182 | 1GB ram free needed, or up to 1GB ram? |
| 14:12.05 | Samot | A 1 CPU x 1GB RAM VM will do it |
| 14:12.27 | BoBeR182 | cool cool, do you know when that server might get overloaded? |
| 14:12.29 | BoBeR182 | 10users |
| 14:12.30 | BoBeR182 | 100? |
| 14:12.42 | Samot | Depends on what you are doing. |
| 14:12.55 | *** join/#asterisk nix8n82 (~AndChat58@67-130-74-235.dia.static.qwest.net) |
| 14:12.57 | BoBeR182 | could you give me some examples? |
| 14:12.58 | Samot | How many calls, queues, playback, recordings... |
| 14:13.08 | Samot | video |
| 14:13.21 | BoBeR182 | No video, strictly Calling |
| 14:13.22 | Samot | Just like any type of system, stuff that's going to take resources.. |
| 14:13.49 | Samot | You can get away with about 10 users if its just calls. |
| 14:13.59 | BoBeR182 | lovely |
| 14:14.18 | BoBeR182 | Thank you for the help, is it worth setting up asterisk normally |
| 14:14.24 | BoBeR182 | or just grabbing the Live CD? |
| 14:14.40 | Samot | All depends.. |
| 14:14.55 | Samot | If you've never done this and just are looking for a PBX solution for a couple phones... |
| 14:15.01 | Samot | Go get FreePBX |
| 14:16.15 | BoBeR182 | just reading about that now... |
| 14:16.24 | BoBeR182 | I have 3 DIDs |
| 14:16.34 | BoBeR182 | but when I do my outbound calls I want them all from the same number |
| 14:16.48 | BoBeR182 | does FreePBX support setting the caller ID? |
| 14:16.55 | Samot | Yes. |
| 14:17.06 | Samot | It's a GUI based distro using Asterisk. |
| 14:17.12 | igcewieling | BoBeR182: Are they analog lines? |
| 14:17.17 | BoBeR182 | Is it more resource heavy |
| 14:17.21 | Samot | No. |
| 14:17.23 | BoBeR182 | igcewieling, no SIP |
| 14:17.50 | igcewieling | BoBeR182: Good. |
| 14:18.17 | *** join/#asterisk nix8n82 (~AndChat58@2600:100e:b025:29ff:6465:6523:38f4:52f4) |
| 14:20.30 | [TK]D-Fender | <Samot> You can get away with about 10 users if its just calls. <- I had 30 users on a 3ghz 1-core Xeon w/ 2 gb easy |
| 14:21.13 | Samot | Probably could... |
| 14:21.20 | Samot | Not saying it's not possible. |
| 14:21.30 | Samot | But it's two users now. |
| 14:21.31 | [TK]D-Fender | BoBeR182, Depends on your provider. Most plans let you set whatever you want. Some may let youset from a fixed list you may have to provide them. Other plans treat each "line" as a separate service with no rights to change the #, etc |
| 14:21.42 | [TK]D-Fender | Samot, Yeah, he's more than fine. |
| 14:21.45 | Samot | I think 1 CPU 1GB RAM is fine. |
| 14:21.59 | [TK]D-Fender | an rPi can handle around 4 calls or so from what I've heard |
| 14:22.02 | BoBeR182 | Cool cool |
| 14:22.11 | BoBeR182 | I'll have it probably in a VM |
| 14:22.47 | *** part/#asterisk BoBeR182 (~BoBeR182@gateway/tor-sasl/bober182) |
| 14:22.56 | VTheThinker | Samot anyway, do you know any irc channel where I can ask about that? somehting more general on this type of devices... |
| 14:23.27 | Samot | Is there #openvox? |
| 14:23.34 | Samot | This is an OpenVox appliance.... |
| 14:23.49 | VTheThinker | no, they donn't have a channel |
| 14:23.57 | Samot | This is an appliance they built.. |
| 14:24.07 | Samot | What it does? Don't know. |
| 14:24.18 | Samot | If you can't access it, asking here isn't the right place. |
| 14:25.21 | Samot | You'll have to go on their site and look for their support channels. |
| 14:25.26 | Samot | Whatever they maybe... |
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| 15:52.48 | *** join/#asterisk kchehab (~Denis19@94.187.48.207) |
| 15:53.02 | kchehab | hi how to store cancel call in asterisk cdrdb |
| 15:53.24 | kchehab | asteriskcdrdb |
| 15:53.49 | kchehab | as if i send a call then cancel ,the call will have no record ... |
| 15:54.02 | kchehab | Asterisk 11.19.0 |
| 16:05.01 | *** join/#asterisk matrix1233 (~matrix123@41.230.38.112) |
| 16:06.12 | kchehab | how to set Log unanswered calls: No to be Log unanswered calls: Yes |
| 16:07.08 | [TK]D-Fender | cdr.conf |
| 16:17.30 | kchehab | yes i find it 10x |
| 16:18.52 | kchehab | [TK]D-Fender this should log in asteriskcdrdb cdr table or cel table ? |
| 16:19.30 | kchehab | the cancled call as i set it to yes and it show enable , but still the cancel call is not presented in cdr table |
| 16:25.26 | *** join/#asterisk matrix1233 (~matrix123@41.230.38.112) |
| 16:26.27 | kchehab | enable=yes |
| 16:26.28 | kchehab | unanswered = yes |
| 16:26.28 | kchehab | endbeforehexten=yes |
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| 19:20.06 | Kobaz | [TK]D-Fender: meep |
| 19:28.59 | *** join/#asterisk CyberJacob (~CyberJaco@bouncer.bluesapphiremedia.co.uk) |
| 19:41.48 | Kobaz | sooooo |
| 19:41.55 | Kobaz | i'm having a weird nat problem |
| 19:42.29 | Kobaz | peer, nat=force_rport,comedia |
| 19:42.37 | Kobaz | that's all fine and dandy |
| 19:43.09 | Kobaz | [2017-06-30 15:25:47.951] Peer audio RTP is at port 192.168.250.29:2230 [2017-06-30 15:25:47.951] Transmitting (NAT) to 10.20.1.22:1024: |
| 19:43.18 | Kobaz | i'll be pasting up the full sip debug in a sec... buuut |
| 19:43.24 | Kobaz | what the hell is going on |
| 19:43.45 | Kobaz | asterisk knows the peer is NAT'd. it is sending NAT'd sip. but not handling NAT on the rtp side! |
| 19:44.07 | Kobaz | Peer audio RTP is supposed to be 10.20.1.22:2230 |
| 19:52.09 | *** join/#asterisk matrix1233 (~matrix123@41.230.38.112) |
| 19:58.36 | salviadud | Kobaz, what is your sip peer? |
| 19:59.07 | Kobaz | 10.20.1.22 |
| 19:59.17 | salviadud | Not what IP, what is it |
| 19:59.20 | salviadud | a sip phone |
| 19:59.23 | salviadud | softphone |
| 19:59.24 | Kobaz | oh |
| 19:59.31 | Kobaz | the 10.20.1.22 is a vpn/router |
| 19:59.34 | Kobaz | and then behind that's a polycom 331 |
| 20:00.11 | salviadud | so, the polycom is behind that firewall, it registers with asterisk, but no audio |
| 20:00.24 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-saqkxzjzkucycpef) |
| 20:00.52 | Kobaz | yeah |
| 20:00.53 | Kobaz | one way audio |
| 20:01.29 | Kobaz | no audio from the server to the client/phone |
| 20:01.40 | Kobaz | since asterisk is using 192.168.250.29:2230 |
| 20:01.46 | Kobaz | which is the subnet *behind* the vpn |
| 20:01.56 | Kobaz | but nat is enabled, soooo |
| 20:02.01 | Kobaz | it shouldn't be using that ip |
| 20:02.02 | salviadud | what about rtp.conf |
| 20:02.44 | Kobaz | what about it? it's pretty basic, just my local port range |
| 20:03.12 | salviadud | just a checklist |
| 20:03.35 | Kobaz | yeah yeah |
| 20:03.49 | salviadud | what version of asterisk is it? |
| 20:03.50 | Kobaz | really nothing fancy. rtpstart=16000 rtpend=17000 |
| 20:03.56 | Kobaz | but that has nothing to do with this leg of the call |
| 20:03.58 | Kobaz | this is rtp -> phone |
| 20:04.09 | Kobaz | the phone/gw's local rtp is 10.20.1.22:2230 |
| 20:04.14 | Kobaz | 1.8 hehe, yeah it's old |
| 20:04.29 | salviadud | even that shouldn't matter too much |
| 20:04.40 | salviadud | but would it kill you to upgrade to 11.25.1 at least? |
| 20:05.02 | salviadud | I'd try it with pjsip |
| 20:05.05 | drmessano | Then what? |
| 20:05.25 | Kobaz | i can't |
| 20:05.43 | Kobaz | on this box, 11.anything the performance is awful and basically nukes the box |
| 20:05.44 | salviadud | have you done a set sip debug on to see what's going on? |
| 20:05.49 | Kobaz | yeah plenty of sip debug |
| 20:05.52 | Kobaz | that's from the sip debug |
| 20:05.55 | Kobaz | lemme paste the whole thing |
| 20:06.06 | lorsungcu | Kobaz: *13 performs worse than 1.8? |
| 20:06.07 | drmessano | waits for "Did you reboot.. 2 or 3 times" |
| 20:06.16 | lorsungcu | somehow i seriously doubt that |
| 20:06.51 | Kobaz | lorsungcu: don't know yet, i haven't migrated our platform to anything newer than 11 |
| 20:07.21 | lorsungcu | Kobaz: 11 is EOL this year |
| 20:07.25 | lorsungcu | you should probably consider that |
| 20:07.52 | Kobaz | https://pastebin.com/XZMpikJR |
| 20:07.56 | Kobaz | oh it's definitely on the list |
| 20:08.00 | Kobaz | it's just a 100 hour project |
| 20:08.17 | lorsungcu | putting it off won't change that |
| 20:08.19 | lorsungcu | :p |
| 20:08.23 | Kobaz | hah yeap |
| 20:08.29 | Kobaz | anyway so you have your fun stuff here |
| 20:08.31 | Kobaz | INVITE sip:6863@192.168.250.29 SIP/2.0 |
| 20:08.38 | Kobaz | and nat=yes on this peer |
| 20:09.39 | Kobaz | [2017-06-30 15:25:47.951] set_destination: set destination to 192.168.250.29:5060 |
| 20:09.55 | drmessano | What are your localnet settings? |
| 20:09.56 | Kobaz | it's using the wrong destination. despite being instructed otherwise |
| 20:09.58 | Kobaz | that's the problem |
| 20:09.59 | Kobaz | it's in there |
| 20:10.04 | Kobaz | scroll down a bit |
| 20:12.22 | drmessano | Yeah nothing seems off there |
| 20:12.37 | Kobaz | that's what i thought |
| 20:12.38 | Kobaz | yeap |
| 20:12.47 | Kobaz | this started happening like a week ago |
| 20:12.59 | drmessano | I never cared for 1.8.. it always seemed like Asterisk's college experimental era |
| 20:13.04 | Kobaz | haha |
| 20:13.07 | drmessano | Like dudes or chicks, didn't matter |
| 20:13.40 | drmessano | 11.x era it had to get a real job |
| 20:14.58 | lorsungcu | Kobaz: sip show peer ? |
| 20:15.26 | Kobaz | 6863/6863 10.20.1.22 D N 1027 OK (63 ms) |
| 20:15.47 | lorsungcu | sip show peer 6863 |
| 20:16.45 | Kobaz | https://pastebin.com/PAWPATwr |
| 20:17.07 | Kobaz | <PROTECTED> |
| 20:17.19 | Kobaz | not sure why but 1.8 doesn't spit out the value of comedia |
| 20:18.06 | lorsungcu | what is it actually set to in the peer? just yes, right? |
| 20:18.14 | *** join/#asterisk jkroon (~jkroon@41.13.56.54) |
| 20:18.16 | Kobaz | i've tried both =yes |
| 20:18.23 | Kobaz | and =force_rport,comedia |
| 20:18.30 | Kobaz | generally on 1.8 i just use yes and it always works |
| 20:18.31 | Kobaz | never had a problem |
| 20:18.37 | lorsungcu | iirc thats what it sohuld be |
| 20:18.42 | Kobaz | yeah |
| 20:18.42 | lorsungcu | =yes |
| 20:20.20 | salviadud | remembers 2010 |
| 20:20.51 | salviadud | That's how old 1.8 is if you put into account when it was released. |
| 20:21.22 | Kobaz | hehe |
| 20:24.58 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 20:25.22 | Kobaz | hey [TK]D-Fender |
| 20:26.41 | drmessano | salviadud: Are you the "Captain Obvious" guy in your circle of people? |
| 20:26.45 | drmessano | Asking for a friend |
| 20:27.34 | salviadud | Not really, I just have this philosophy that states "never expect a different result, trying the same thing" |
| 20:28.00 | [TK]D-Fender | I used to be "Captain Obvious" until I got my promotion to "Major Buzzkill"... |
| 20:28.09 | Kobaz | [TK]D-Fender: soooo wanna do some sips? |
| 20:32.45 | igcewieling | General Disarray 8-| |
| 20:34.51 | Kobaz | [TK]D-Fender: mind taking a look at a sip issue? |
| 20:35.12 | [TK]D-Fender | You know the drill... |
| 20:35.14 | *** join/#asterisk miralin (~Thunderbi@91.237.94.1) |
| 20:35.19 | Kobaz | https://pastebin.com/XZMpikJR |
| 20:38.53 | *** join/#asterisk overyander (~jeff@12.49.160.131) |
| 20:40.02 | Kobaz | [TK]D-Fender: sooooo |
| 20:40.11 | Kobaz | [TK]D-Fender: why is rtp -> 192.168.250.29 |
| 20:40.19 | Kobaz | even though NAT is enabled/used |
| 20:41.37 | [TK]D-Fender | FULL call |
| 20:41.48 | Kobaz | that is the full call setup |
| 20:41.55 | Kobaz | that's the asterisk dialplan, and the sip invite/ok/etc |
| 20:45.12 | [TK]D-Fender | FIRST damn leg <---- |
| 20:45.20 | Kobaz | why? it's not related |
| 20:45.30 | Kobaz | the problem is media is sending to 192.168.250.0 |
| 20:45.44 | Kobaz | the first leg works perfectly |
| 20:45.58 | Kobaz | i can show it to you if you really want to, but it's just call setup between a soft phone and asterisk |
| 20:46.11 | Kobaz | and media is flowing perfectly there |
| 20:46.30 | *** join/#asterisk matrix1233 (~matrix123@41.230.38.112) |
| 20:58.59 | Kobaz | [TK]D-Fender: any idea? |
| 21:12.34 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 21:13.35 | Kobaz | [TK]D-Fender: it looks like asterisk is not respecting nat=yes on the RTP |
| 21:14.43 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 21:19.51 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
| 21:26.38 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 21:33.01 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 21:49.13 | *** join/#asterisk matrix1233 (~matrix123@41.230.38.112) |
| 21:55.28 | *** join/#asterisk wabbits (~rtreleave@ip-64-140-118-201.user.start.ca) |
| 21:57.20 | *** join/#asterisk matrix1233 (~matrix123@41.230.38.112) |
| 22:00.22 | Kobaz | [TK]D-Fender: poke :) |
| 22:09.03 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 22:31.22 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 22:37.32 | *** join/#asterisk matrix1233 (~matrix123@41.230.38.112) |
| 22:48.56 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 22:54.07 | *** join/#asterisk kchehab (~Denis19@94.187.48.207) |
| 22:56.16 | kchehab | i cant store the cancel transaction into asteriskcdr db , i set cdr.conf it unanswered = yes ( Logging: Enabled) |
| 22:56.57 | kchehab | is there a way to save the cancel 487 Request Terminated | in cdr ?? |
| 23:00.39 | [TK]D-Fender | that is a SIP message |
| 23:00.58 | [TK]D-Fender | nothing about that says WHAT the request was or if it was even rlated to a call |
| 23:01.12 | [TK]D-Fender | You haven't described the call being placed or shown it yet |
| 23:01.29 | *** part/#asterisk kharwell (kharwell@nat/digium/x-gfqtiychwlyhopaz) |
| 23:16.59 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-170.cust.bezeqint.net) |
| 23:30.47 | *** join/#asterisk kchehab (~Denis19@94.187.48.207) |
| 23:30.49 | kchehab | all signalling are ok just the cdr issue |
| 23:43.55 | kchehab | call go to [h@dialter:1 |
| 23:44.04 | kchehab | as i call it from another contect |
| 23:44.10 | kchehab | contect |
| 23:44.15 | kchehab | context |
| 23:45.28 | kchehab | i will check hangupcause :) |
| 23:52.44 | kchehab | the hangupcause is always 0 |