| 00:04.00 | darkunderlord_ | isn't the PJSIP stuff the "right" way to do it? |
| 00:04.41 | file | it's a way, whether it is right or not depends on the situation and deployment |
| 00:04.46 | drmessano | I don't see how the XMPP way is "old and clunky" |
| 00:07.19 | [TK]D-Fender | PJSIP just offered another way |
| 00:07.29 | [TK]D-Fender | And didn't exist until * 12 |
| 00:08.59 | [TK]D-Fender | (chan_pjsip that is) |
| 00:10.09 | darkunderlord_ | if XMPP isn't considered old, I'll try that. I just want to be as future proof as possible |
| 00:11.17 | [TK]D-Fender | Or you could try doing it properly with JPSIP |
| 00:11.19 | [TK]D-Fender | your choice |
| 00:11.45 | drmessano | ..... |
| 00:11.49 | [TK]D-Fender | Feel free to describe what makes one method more future-proof than the other.... |
| 00:12.01 | drmessano | I don't understand the "old" metric |
| 00:12.19 | [TK]D-Fender | drmessano, Sorry, we're devoid of qualifiers here ;) |
| 00:12.22 | drmessano | Asterisk is basically "old" |
| 00:12.40 | lorsungcu | darkunderlord_: has anything you've done *worked*? |
| 00:12.42 | file | you can also add an anonymous endpoint and do it like you were doing with chan_sip |
| 00:13.28 | drmessano | "old" > "not worked" |
| 00:15.22 | lorsungcu | sounds to me like old = widely aopted. |
| 00:15.51 | lorsungcu | drmessano: dream something up that uses node. thats what hes afte.r |
| 00:16.02 | drmessano | Oh |
| 00:16.05 | drmessano | Well |
| 00:16.18 | drmessano | He could use notify.js |
| 00:16.32 | drmessano | It just takes 3 days to compile |
| 00:16.36 | lorsungcu | ok perfect |
| 00:16.44 | darkunderlord_ | well chan_sip was around for quite a while. Now it's not the standard going forward. Some things are only community supported and others are digium supported. |
| 00:16.50 | darkunderlord_ | So I'm thinking in that realm |
| 00:17.11 | drmessano | Wrong realm |
| 00:17.18 | drmessano | Lets be honest |
| 00:17.27 | darkunderlord_ | I don't think so. |
| 00:17.36 | drmessano | If you deployed a new server today with chan_sip |
| 00:17.43 | drmessano | no PJSIP at all |
| 00:17.47 | drmessano | Your phones would work |
| 00:17.50 | drmessano | in 5 years |
| 00:17.54 | drmessano | Your phones would work |
| 00:18.00 | drmessano | There is no OS rot |
| 00:18.06 | [TK]D-Fender | darkunderlord_, Ok, even on that premise : my tell us between XMPP & PJSIP, which one is it? |
| 00:18.08 | drmessano | This is purely FOMO |
| 00:18.20 | drmessano | or plain old FOTO |
| 00:18.27 | drmessano | Sorry |
| 00:18.30 | drmessano | FOTU |
| 00:18.52 | darkunderlord_ | have you ever purchased a digium support contract? |
| 00:18.59 | darkunderlord_ | it does make a difference |
| 00:19.00 | [TK]D-Fender | darkunderlord_, Which is on the chopping block? |
| 00:19.08 | drmessano | Never needed to.. I support my own boxes |
| 00:19.17 | darkunderlord_ | lol. |
| 00:19.39 | drmessano | Maybe you should just buy a SwitcVOX |
| 00:19.44 | lorsungcu | darkunderlord_: that is quite a change from "thats old" to "digium wont support it" |
| 00:19.46 | drmessano | Maybe you should just buy a SwitchVOX |
| 00:19.51 | darkunderlord_ | I typica,ly do to, but if you ask file about these kinds of things even he will tell you to stick to certain things |
| 00:19.55 | darkunderlord_ | OMG, seriously? |
| 00:20.05 | darkunderlord_ | F' switchvox |
| 00:20.07 | lorsungcu | call digium, ask what is best supported, or will be for the longest |
| 00:20.08 | lorsungcu | go with that |
| 00:20.10 | drmessano | Yeah or a nice Grandstream appliance |
| 00:20.11 | darkunderlord_ | I've been using asterisk since v1 |
| 00:20.13 | lorsungcu | asking on IRC will get you precisely this |
| 00:20.24 | drmessano | Grandstream makes a nice PBX |
| 00:20.25 | [TK]D-Fender | <[TK]D-Fender> darkunderlord_, Ok, even on that premise : my tell us between XMPP & PJSIP, which one is it? |
| 00:20.27 | darkunderlord_ | if you don't wan tto help someone, dont' do to linux forums or irc |
| 00:20.31 | [TK]D-Fender | <[TK]D-Fender> darkunderlord_, Which is on the chopping block? |
| 00:20.33 | [TK]D-Fender | well? |
| 00:20.53 | darkunderlord_ | I'd rather do what the newer way is, because it'll be around longer most likely |
| 00:21.09 | darkunderlord_ | and I'm using realtime, so if that helps with PJSIP, I'd rather control it in my code from the DB |
| 00:21.10 | drmessano | darkunderlord_: "lack of desire to help" != "demanding rational thought" |
| 00:21.26 | [TK]D-Fender | darkunderlord_, You also don't want help. We've kinda established that already |
| 00:21.35 | darkunderlord_ | lol |
| 00:21.41 | darkunderlord_ | I'll see you guys at astricon, yet again |
| 00:22.04 | drmessano | No you won't |
| 00:22.32 | drmessano | I'll be busy working.. Don't have time to rub elbows and brag about what I know |
| 00:22.33 | [TK]D-Fender | ditto |
| 00:22.33 | darkunderlord_ | If you aren't helping, then I'll just keep banging my head. |
| 00:22.38 | darkunderlord_ | hahahahaha |
| 00:22.44 | [TK]D-Fender | [TK]D-Fender> darkunderlord_, You also don't want help. We've kinda established that already <- |
| 00:23.14 | darkunderlord_ | Fender, you've been a dick most of the day to anyone who needs help in here. Not just to me |
| 00:23.23 | [TK]D-Fender | darkunderlord_, Not here |
| 00:23.28 | [TK]D-Fender | And "oh please" |
| 00:23.29 | drmessano | lol |
| 00:23.30 | darkunderlord_ | agreed |
| 00:23.40 | [TK]D-Fender | I've had my time wasted by mororn who won't ready what you tell them 5 times |
| 00:23.45 | [TK]D-Fender | and can't follow instructions. |
| 00:23.54 | [TK]D-Fender | I am in no way an outright "dick". |
| 00:24.06 | darkunderlord_ | you're just another RTFM guy. I've been doign that for years |
| 00:24.07 | lorsungcu | he's actually very cuddly |
| 00:24.08 | [TK]D-Fender | I've been ticked off for people wasting my time, nothig more |
| 00:24.17 | darkunderlord_ | Noone makes you hang out here |
| 00:24.32 | darkunderlord_ | I'm only here because I need help |
| 00:24.51 | [TK]D-Fender | darkunderlord_, Ok, fine. You showed NO debug. You showed no configs. y ou asked NO details on how any of it wasa supposed to work. You asked NOTHING about getting your shit to work |
| 00:24.59 | [TK]D-Fender | YOU DIDN'T GIVE ANYTHING FOR US TO HELP YOU WITH |
| 00:25.00 | *** join/#asterisk bhans (~bhans@unaffiliated/bhans) |
| 00:25.02 | drmessano | Just that it cant be "old" |
| 00:25.12 | [TK]D-Fender | SSo if you don't SHOW somethting and don't ask.. then you clearly don't want help[ |
| 00:25.38 | [TK]D-Fender | darkunderlord_> I'm only here because I need help <- Where's something to LOOK at to help you? You never gave anything |
| 00:25.41 | [TK]D-Fender | So don't go bitching |
| 00:25.43 | drmessano | darkunderlord_: Also, the channel is logged.. feel free to review TK's 12+ years of IRC help. You're kind making this personal and speaking out of your ass |
| 00:25.51 | drmessano | kinda* |
| 00:26.20 | drmessano | No reason to pick a fight because your plans are being questioned |
| 00:26.59 | [TK]D-Fender | Then you topic jumped on the fact you missed tha page offering alternative. Said it didn't. I confirmed otherwise, then you complained about other methods being old once I proved they were even there when you didn't see them yourself |
| 00:27.18 | drmessano | "I want BLF/hints between servers and it cant be some old shit" <--- Thats all we know |
| 00:27.31 | [TK]D-Fender | If you wanted help you'd have shown us what was failing. |
| 00:27.35 | drmessano | and a whole bunch of "hahahahhaha" |
| 00:27.57 | [TK]D-Fender | Saying what you want to accomplish != asking for and providing something to be helped WITH |
| 00:28.40 | drmessano | If you have a Digium support contract, maybe contact them and ask them what solution will be supported in 5 years |
| 00:28.57 | drmessano | Because we're just a bunch of dumbasses that use what is available |
| 00:29.03 | [TK]D-Fender | reloads res_psychic.so |
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| 00:29.19 | [TK]D-Fender | That sucker has been mothballed for ages! |
| 00:33.01 | drmessano | I guess also going to Astricon makes you more authoritative on all things Asterisk. |
| 00:33.39 | [TK]D-Fender | no, it doesn't |
| 00:33.39 | drmessano | That is some serious entitlement |
| 00:33.45 | [TK]D-Fender | I've NEVER been |
| 00:33.49 | [TK]D-Fender | I can read docs |
| 00:33.52 | [TK]D-Fender | I look at debug |
| 00:34.01 | [TK]D-Fender | I have a functioning brain |
| 00:34.03 | drmessano | Obviously you have never been |
| 00:34.05 | drmessano | We can tell. |
| 00:34.12 | drmessano | You're oozing with dumbass |
| 00:34.26 | drmessano | #sarcasm |
| 00:34.31 | drmessano | #notdead |
| 00:35.02 | [TK]D-Fender | And I've been using * for over 13 years now before they hit a solid integer. |
| 00:35.15 | drmessano | lorsungcu: please add "Have you ever been to Astricon?" to the toolkit, please |
| 00:35.18 | [TK]D-Fender | And you've still shown nothing and continue to whine |
| 00:35.24 | [TK]D-Fender | as I said, you don't want help |
| 00:35.34 | [TK]D-Fender | Otherwise we'd have seen configs & debug ages ago |
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| 01:03.27 | [TK]D-Fender | #crickets |
| 01:06.46 | drmessano | Probably went to PM |
| 01:14.57 | darkunderlord_ | I just helped with finding a bug in the configure script for corosync, having never used corosync before. I do try, I do research. So..anyone have a favorite jabber server then? Ejabberd, openfire, etc? |
| 01:18.51 | drmessano | I believe Prosody implements enough of XEP-0060 to work.. and it's easy to set up |
| 01:19.23 | darkunderlord_ | thx |
| 01:21.50 | darkunderlord_ | sorry, I took offense. I'm a huge fan and LOVE asterisk. Used it a long time and have stickers on all my ownership, including last two cars. I'll try to ask smarter questions. |
| 01:26.48 | [TK]D-Fender | Well you've already abandoned the method you started with |
| 01:26.56 | [TK]D-Fender | and we never got a chance to get that to work |
| 01:28.16 | [TK]D-Fender | So when we've got something worth looking at I might bother to. |
| 01:28.26 | [TK]D-Fender | heads off to work out for a while |
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| 01:37.38 | darkunderlord_ | I really like the way file had me going, but when identifying overrode my valid clients, and I'm trying to read up on the anonymous stuff now. I'll check back when i hopefully find enough info that I can ask a good question. |
| 02:21.47 | darkunderlord_ | wow. Finally got it working. Thanks and sorry to all. |
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| 10:03.30 | pagios | hi what is the best hardware to buy to convert my desktop into a pbx? |
| 10:04.26 | Kunsi | you'll need a mainboard, a cpu, some ram and a hdd/ssd |
| 10:04.46 | Kunsi | bonus points for some hardware to connect phones |
| 10:07.10 | pagios | Kunsi, i need to connect my phone line to my desktop, so what hardware can i use |
| 10:08.43 | Kunsi | what kind of line? analog, isdn, sip, $other? |
| 10:10.50 | pagios | Kunsi, analog |
| 10:10.52 | pagios | PSTN |
| 10:11.23 | pagios | i mainly want to have a automatic greeting with extensions, and skype connect integration for users to dial in and for me to dial out using my phone |
| 10:11.26 | Kunsi | i got a linksys spa-3102 analog-to-sip converter-thingie, works fine |
| 10:20.27 | pagios | it can do all the above Kunsi ? |
| 10:21.03 | Kunsi | it converts analog to sip. asterisk is doing some magic then |
| 10:21.13 | pagios | Kunsi, this?https://www.amazon.com/Cisco-SPA3102-Voice-Gateway-Router/dp/B000FKP55U |
| 10:24.16 | Kunsi | test is correct, but image looks wrong. should look like https://sc02.alicdn.com/kf/HTB18_52KpXXXXbSXFXXq6xXFXXXM/Linksys-SPA-3102-SPA-3102-NA-VoIP.jpg |
| 10:27.54 | pagios | 200usd |
| 10:27.59 | pagios | any cheap pci card for instance? |
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| 12:28.18 | [TK]D-Fender | SPA-3102 is NOT 200$USD |
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| 13:21.41 | darkunderlord_ | pagios, I think you need to make sure whatever you get, it does FXO at least. Not a cheaper FXS only device. |
| 13:22.00 | pagios | FXO and FXS |
| 13:22.01 | pagios | 1x 1 |
| 13:22.07 | pagios | do you recommend any pci chip |
| 13:23.57 | [TK]D-Fender | All of those are generally pricier or flakey |
| 13:24.34 | darkunderlord_ | I agree with getting the SPA-3102 |
| 13:24.59 | darkunderlord_ | https://www.amazon.com/GrandStream-HT503-1-FXS-Analog-Telephone/dp/B002H29TGA/ref=sr_1_1?ie=UTF8&qid=1500902684&sr=8-1&keywords=spa3102 |
| 13:25.04 | darkunderlord_ | 50 bucks on amazon |
| 13:25.53 | Samot | You paid 200USD for a SPA3102? You were ripped off. |
| 13:26.02 | [TK]D-Fender | ~gs |
| 13:26.03 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 13:27.19 | Kunsi | pagios: where are you from? i'm not using my spa3102 anymore, would sell it (but maybe shipping to you would be expensive) |
| 13:29.46 | pagios | darkunderlord_, so mainly asterisk would be running on that grandstream ht503 you mean or need to install asterisk on my pc? |
| 13:30.20 | darkunderlord_ | wtf, I didn't select the grandstream. lol Not sure why amazon gave me that link. Don't get that. :D |
| 13:30.29 | darkunderlord_ | the SPA3102 |
| 13:30.39 | pagios | darkunderlord_, idea is i want to control asterisk on my desktop |
| 13:30.43 | pagios | much more flexibility then |
| 13:30.51 | pagios | i just need to get my phone line into my desktop machine |
| 13:30.57 | pagios | and install asterisk on a vm |
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| 13:31.06 | darkunderlord_ | pagios, yeah I know. that's why you should get the 3102 |
| 13:31.21 | pagios | darkunderlord_, is there like any usb thing? |
| 13:31.32 | pagios | where i can get 1fxo 1 fxs port |
| 13:31.36 | darkunderlord_ | I've never used or heard of that. |
| 13:31.40 | pagios | that way i can link it to my VM |
| 13:31.48 | pagios | coz asterisk in the vm needs to see the dongle |
| 13:31.51 | pagios | or pci etc... |
| 13:32.07 | darkunderlord_ | yeah, use the 3102 man. It's the best option. |
| 13:32.15 | Kunsi | SIP is the way to go. |
| 13:32.24 | darkunderlord_ | I use them for things like paging, etc too for work. |
| 13:32.26 | pagios | how do you share the 3102 with the vm? |
| 13:32.36 | darkunderlord_ | it's a SIP client/endpoint |
| 13:32.36 | [TK]D-Fender | <pagios> darkunderlord_, idea is i want to control asterisk on my desktop <- what does this mean? |
| 13:32.51 | darkunderlord_ | [TK]D-Fender, agree, that sounded odd. |
| 13:32.51 | Kunsi | it's ethernet, just connect it to the same switch |
| 13:32.58 | pagios | [TK]D-Fender, it means i want to make my desktop as the asterisk server |
| 13:33.13 | [TK]D-Fender | Why? |
| 13:33.23 | [TK]D-Fender | You intending to leave that on 24/7? |
| 13:33.24 | pagios | to get more Hardisk space, more flexibility |
| 13:33.26 | pagios | yea |
| 13:33.28 | pagios | exactly |
| 13:33.40 | pagios | i can then integrate with some scritps on my desktop |
| 13:33.41 | pagios | etc |
| 13:33.44 | [TK]D-Fender | HD space is practically irrelevant |
| 13:34.02 | darkunderlord_ | pagios, so you already run Linux on your desktop? Or you'll run it in a VM or something? |
| 13:34.06 | pagios | yea |
| 13:34.13 | pagios | i will run it in a VM |
| 13:34.16 | darkunderlord_ | yeah asterisk doesn't need much HD sapce |
| 13:34.19 | pagios | and that vm will stay 24/7 |
| 13:34.37 | [TK]D-Fender | I'm using 13% on an 80GB HD installed in 2012 for my company's PBX |
| 13:35.02 | [TK]D-Fender | You can do it if you want, but that's a machine you'd be doing other stuff with which isn't great |
| 13:35.05 | [TK]D-Fender | but whatever |
| 13:35.16 | pagios | its a powerful machine |
| 13:35.28 | pagios | but the thing is i need to share the hardware wit hthe VM |
| 13:35.33 | darkunderlord_ | just spend the 50 bucks on the 3102. There is alot of documentation on how to get it connected to Asterisk. |
| 13:35.35 | pagios | so i need a usb based hardware with fxo |
| 13:35.56 | [TK]D-Fender | SPA = ethernet |
| 13:35.59 | darkunderlord_ | that's even more reason to use a separate device that talks SIP. |
| 13:36.00 | [TK]D-Fender | best option |
| 13:36.12 | darkunderlord_ | sharing hardware wiht VM's is sketchy at best. |
| 13:36.14 | pagios | darkunderlord_, oh you mean integrate the 3102 with asterisk over ethernet |
| 13:36.21 | [TK]D-Fender | that's what it does |
| 13:36.25 | darkunderlord_ | pagios, that's what all of us mean ;) |
| 13:36.31 | pagios | i thought asterisk runs on the 3102 |
| 13:36.37 | [TK]D-Fender | You should have read up on it when it was first mentioned |
| 13:36.38 | pagios | haha |
| 13:36.39 | [TK]D-Fender | no. |
| 13:36.40 | darkunderlord_ | nope, it's just the way of talking to the analog |
| 13:36.43 | pagios | ok i get it now |
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| 13:36.56 | darkunderlord_ | good |
| 13:37.00 | [TK]D-Fender | the SPA-3102 takes in 1 phones, and 1 line and sends them over SIP to your server |
| 13:37.21 | pagios | so the flow is PSTN -> FXO -> 3102 -> LAN -> Asterisk on my Desktop -> Ibterbet |
| 13:37.27 | darkunderlord_ | that will be your "trunk" for outbound and inbound calls. |
| 13:37.32 | pagios | right? |
| 13:37.49 | darkunderlord_ | pagios, pretty much. The 3102 IS the FXO though. |
| 13:38.08 | pagios | and it can be the otherway around |
| 13:38.14 | darkunderlord_ | yep |
| 13:38.24 | pagios | internet >- asterisk server on my vm -> lan -> 3102 -> pstn |
| 13:38.33 | pagios | then i can bridge mode my vm with the lan |
| 13:38.37 | pagios | and thats it |
| 13:38.42 | darkunderlord_ | definitely. Now you got it. |
| 13:38.47 | pagios | hehe ok |
| 13:38.48 | pagios | thanks :) |
| 13:39.15 | pagios | <PROTECTED> |
| 13:39.26 | darkunderlord_ | where? that's way overpriced |
| 13:39.54 | pagios | darkunderlord_, https://www.amazon.com/gp/offer-listing/B000FKP55U/ref=dp_olp_0?ie=UTF8&condition=all |
| 13:40.30 | Kunsi | pagios: where are you from? i'm not using my spa3102 anymore, would sell it (but maybe shipping to you would be expensive) |
| 13:40.39 | Kunsi | (asking again) |
| 13:40.40 | darkunderlord_ | maybe something else has replaced the SPA3102? |
| 13:40.50 | pagios | thanks but prefer to get a new one :/ |
| 13:41.12 | Samot | The SPA3102 is not worth $200USD even new |
| 13:41.15 | Samot | It's OLD |
| 13:41.18 | pagios | whats new |
| 13:41.20 | Samot | Like 10+ years. |
| 13:41.30 | Samot | Getting a new one is fine, will be hard, but fine |
| 13:41.35 | pagios | need a low cost one |
| 13:41.40 | Samot | But even new, sealed in a box, is not worth $200USD |
| 13:41.49 | pagios | i was thinking of a pci adapter or usb if available |
| 13:41.56 | [TK]D-Fender | pagios, Where are you located? |
| 13:42.01 | darkunderlord_ | 50 bucks on ebay, or get it from Samot |
| 13:42.14 | pagios | Ireland |
| 13:43.02 | [TK]D-Fender | e-bay is your best bet on this so far |
| 13:43.51 | pagios | i need an alternative one |
| 13:43.53 | pagios | that is pretty old |
| 13:43.54 | Samot | I don't have any |
| 13:44.11 | Samot | pagios: The SPA3102's work just fine. |
| 13:44.22 | Kunsi | don't know if anyone is producing devices with FXO ports anymore |
| 13:44.26 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 13:44.32 | Kunsi | everything's VoIP nowadays |
| 13:45.11 | darkunderlord_ | the spa3102 is still your best option. FXO is hella old itself. |
| 13:45.32 | darkunderlord_ | maybe the question is, can you port your analog number to a SIP provider? |
| 13:45.43 | Kunsi | (you also could just check if your router already offers SIP |
| 13:45.48 | Kunsi | -( |
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| 13:48.08 | darkunderlord_ | if you can get off analog, that's your best bet. |
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| 15:40.56 | lvlinux | I need a bit of assistance here. I have in my pjsip.conf both a udp and tcp transport setup. I have one endpoint set to use the tcp transport, but when I do a SIP debug, it shows that Asterisk is continuing to try to talk to the endpoint with UDP, both for registration and calls (asterisk registers to the endpoint). I had it setup with UDP before and it worked fine. I have trasnport=sip-tcp (the name of |
| 15:41.00 | *** join/#asterisk bl3nto (~bl3nto@cpe-109-60-14-187.st3.cable.xnet.hr) |
| 15:41.02 | lvlinux | my tcp transport) in the endpoint config. |
| 15:42.03 | lvlinux | And of course I restarted asterisk after making the changes (pjsip reload doesn't reload transport changes). |
| 15:51.50 | file | the SIP URI controls the transport used |
| 15:52.05 | file | if you want to explicitly use TCP then it would be sip:bob.com\;transport=tcp |
| 15:52.35 | file | not specifying a transport has it follow normal SIP server resolution, so if you give a hostname it'll do an SRV lookup and go from there |
| 15:52.37 | lvlinux | ah ok. |
| 15:53.20 | lvlinux | Oh well that makes sense I guess. But then what is the point of putting transport=tcp in the endpoint section then? |
| 15:53.45 | lvlinux | or rather transport=name-of-defined-tcp-transport |
| 15:54.30 | file | the transport option explicitly uses that transport, in the case of UDP it'll force it to use a specific one |
| 15:55.01 | file | generally you shouldn't need to specify it |
| 15:55.30 | lvlinux | oh, so if I had different NAT parameters on different transport sections or something like that is when that is necessary to specify? |
| 15:56.01 | file | if you have multiple interfaces it tries to be intelligent and use the right transport and IP address information, but it may not get it right |
| 15:56.07 | file | so the transport option allows you to be explicit |
| 15:56.14 | lvlinux | k |
| 15:57.29 | lvlinux | Now, so I should add the \;transport=tcp to the end of all sip:whatever lines, so I need it on server_uri and client_uri parameters in the registration section, and also in the aor section on contact= right? |
| 15:58.45 | file | server_uri you do, contact= yes |
| 15:59.19 | lvlinux | but not client_uri? |
| 15:59.34 | file | not needed, probably won't hurt |
| 15:59.38 | lvlinux | k |
| 16:02.27 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:05.10 | lvlinux | The reason I've switched this endpoint over to TCP is because when it was on UDP, and the server was down, and * tried to make a call with it, * waited quite a long time for the UDP SIP invite to timeout before going on to the next provider. So TCP seems to take care of that. Does that seem like a valid thing to do or are there some timers with UDP that can be messed with that would be preferable than |
| 16:05.17 | lvlinux | mixing transport types? |
| 16:07.32 | igcewieling | enable qualify on the device and stop making things overly complicated |
| 16:08.24 | lvlinux | aha! i totally forgot about qualify! |
| 16:08.39 | lvlinux | thanks igcewieling |
| 16:09.58 | lvlinux | and thanks file for the tcp help! |
| 16:11.20 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 16:24.54 | lvlinux | hmmm, when I turn qualify on and kill the endpoint, asterisk reports it as being offline after 60 seconds but still tries to send calls to it. ?? What am I missing? |
| 16:25.59 | *** join/#asterisk tripleslash (~triplesla@unaffiliated/imsaguy) |
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| 16:29.14 | igcewieling | I've never seen that happen. |
| 16:29.37 | igcewieling | perhaps a pastebin of the cli output showing the issue? |
| 16:29.59 | igcewieling | You're not doing something stupid like running FreePBX? |
| 16:30.14 | lvlinux | NOOOOOOO |
| 16:30.34 | lvlinux | Could it be because the "endpoint" is on localhost? Accessed via 127.0.0.2. |
| 16:30.42 | igcewieling | Dial should immediatly return with a dialstatus of CONTESTION or CHANUNAVAIL |
| 16:31.08 | lvlinux | that's what I thought, but it goes ahead and tries the call and sends 4 invites. Then it waits for timeout. |
| 16:31.11 | igcewieling | Ah, I've never tried running the endpoint and asterisk on the same server. |
| 16:31.40 | igcewieling | running chan_sip or chan_pjsip? |
| 16:31.43 | lvlinux | pjsip |
| 16:32.18 | lvlinux | the "endpoint" is yate running on the same box as asterisk. |
| 16:32.24 | igcewieling | ah, nevermind then. you'll have to ask someone else how qualify works with dial when using pjsip. on my systems (using chan_sip), dial immediatly returns if the endpoint is unreachagle or lagged. |
| 16:32.45 | lvlinux | I would think it should work the same way, but idk. |
| 16:33.18 | lvlinux | When qualify doesn't get a response it comes back with the enpoint is unreachable so it appears to be working, but it still tries the call. |
| 16:33.28 | igcewieling | I'd run a "sip show peer X" where X is the peer name and see what the status is shown as. |
| 16:33.56 | lvlinux | in my case that would be "pjsip show endpoint X" ... lemme do that and check |
| 16:34.50 | file | that was recently made so it behaves that way, older versions won't |
| 16:35.14 | file | as of 13.17.0 and 14.6.0 |
| 16:35.48 | igcewieling | file: specific to pjsip? |
| 16:35.52 | file | yes |
| 16:36.12 | igcewieling | Glad I didn't listen to all those people telling me to upgrade from Asterisk 11 |
| 16:36.30 | igcewieling | Hear that Samot!? |
| 16:36.46 | lvlinux | igcewieling: you can still run chan_sip with 13/14, and the old commands work fine with it. |
| 16:37.00 | Samot | I never said "Go to PJSIP" |
| 16:37.02 | lvlinux | anyway it shows the endpoint as being unavailable just like it should. |
| 16:37.05 | Samot | So.. |
| 16:37.08 | Samot | Irrelevant. |
| 16:37.27 | lvlinux | And PJSIP rocks btw |
| 16:37.50 | igcewieling | I'm sure it does as long is you don't encounter something not supported or not documetned. |
| 16:38.13 | lvlinux | haha like that never happened with chan_sip? |
| 16:38.27 | Samot | Sure. |
| 16:38.29 | Samot | When it was 5 |
| 16:38.30 | igcewieling | lvlinux: not nearly as often |
| 16:38.39 | Samot | But it's not anymore |
| 16:39.01 | lvlinux | That happens with both of them, but the neatness and features of pjsip more than make up for any small drawbacks here and there. |
| 16:39.14 | Samot | In a few years PJSIP will be just like chan_sip in regards to documentation, etc |
| 16:39.28 | Samot | But then the next "new' thing will be out and we'll be having this discussion again. |
| 16:39.37 | lvlinux | lol i hope not |
| 16:39.43 | file | we aren't replacing the SIP stack again |
| 16:39.48 | Samot | I know. |
| 16:39.59 | file | and SIP itself isn't going away for awhile |
| 16:40.14 | DanQuinney | file sounds like he'd kill someone if they changed it again |
| 16:40.22 | Samot | Right. By the time PJSIP is "mature" like Chan_SIP.. |
| 16:40.30 | file | it'd be a waste of time |
| 16:40.33 | file | Samot: define mature |
| 16:41.07 | Samot | More stable development. |
| 16:41.08 | lvlinux | i thought pjsip was already mature---I think Samot meant "old and universally accepted" |
| 16:41.13 | Samot | Right |
| 16:41.17 | Samot | Everyone is using it |
| 16:41.23 | file | ah |
| 16:41.28 | Samot | It's been beaten to the ground in use cases |
| 16:41.35 | Samot | Well documented |
| 16:41.38 | Samot | Supported. |
| 16:41.56 | Samot | The next new SIP stack will have this same thread. |
| 16:43.03 | lvlinux | Sooooo, anyone have ideas about why Asterisk is still trying to dial with my qualified/unavailable endpoint? |
| 16:43.12 | file | lvlinux: I believe I already answered that. |
| 16:43.33 | file | as of 13.17.0 and 14.6.0 it will, if you are using older versions it won't behave that way |
| 16:43.48 | file | if you are using one of those versions then it'd be a bug and an issue should be filed with configuration, console output, etc |
| 16:44.09 | igcewieling | pjsip itseld might be mature, pjsip integration to Asterisk is not mature. |
| 16:44.19 | lvlinux | file: oh! i thought you meant something else when you said that. Sorry |
| 16:44.26 | Samot | How can it be qualified but unavailable? |
| 16:44.32 | Samot | Did I miss something? |
| 16:44.45 | lvlinux | I meant qualify was enabled. |
| 16:45.10 | lvlinux | not that it was "qualified" as in responded to the qualify packet. |
| 16:45.16 | Samot | What do you mean, still send calls to it? |
| 16:45.21 | lvlinux | yes |
| 16:45.35 | Samot | You mean that is still does Dial(SIP/XX) |
| 16:45.52 | Samot | It should. |
| 16:46.02 | Samot | Then it will return a "Subscriber Not Found" |
| 16:46.10 | Samot | So you can handle the call accordingly.. |
| 16:46.13 | Samot | Like send it to VM. |
| 16:46.19 | lvlinux | Yes, and then waits for timeout rather than returning something that allows the call to be handled. So I guess now I have the choice between updating to 13.17 or switching it back to TCP |
| 16:46.19 | Samot | Or dial a "failover" |
| 16:46.27 | Samot | Ahhh. |
| 16:46.30 | Samot | That's what I'm missing. |
| 16:46.36 | salviadud | Is it possible to run chan_sip and chan_pjsip at the same time? |
| 16:46.37 | Samot | It still is dialing per the timeout. |
| 16:46.42 | Samot | Yes. |
| 16:46.47 | lvlinux | salviadud: yes |
| 16:46.48 | Samot | They listen on different ports. |
| 16:47.05 | salviadud | Good to know |
| 16:47.24 | lvlinux | lol that sounds like an administrative nightmare |
| 16:47.35 | Samot | Why? |
| 16:47.50 | Samot | Chan_SIP 5060, PJSIP 5160 |
| 16:47.56 | Samot | They still use the same RTP |
| 16:48.03 | Samot | Just different signaling. |
| 16:48.06 | lvlinux | just keeping track of which you were using for what, debugging etc. |
| 16:48.22 | Samot | It's not that hard. |
| 16:48.45 | lvlinux | Nothing technically bad about it (no different than running IAX and PJSIP simultaneously), but just keeping track of more config files/settings/etc. |
| 16:49.06 | Samot | OK. |
| 16:50.25 | Samot | I should try that approach. |
| 16:50.35 | *** join/#asterisk juvenal (~juvenal@177.197.88.89) |
| 16:50.48 | Samot | "Sorry guys, we can't implement this. It's beyond my config count." |
| 16:51.13 | Samot | "Should have go to me yesterday before I installed PJSIP. That put it to the limit" |
| 16:51.17 | *** join/#asterisk mmlj4 (~mmlj4@47-48-196-90.static.gwnt.ga.charter.com) |
| 16:51.23 | lvlinux | LOL. I guess I'm just gonna switch my endpoint back to TCP. Simpler than upgrading right now... need to upgrade anyway but I've been kindof scared to lol. And I'd at least like to be on the premesis when I upgrade. |
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| 17:41.58 | *** join/#asterisk egassem (86237344@gateway/web/freenode/ip.134.35.115.68) |
| 17:42.01 | egassem | HI sir |
| 17:42.06 | egassem | hows everyone |
| 17:42.59 | egassem | i wanted to ask is it possible to run asterisk where 3 E1 connections are configured with libpri and the fourth as chan_ss7 in the same instense |
| 17:45.00 | [TK]D-Fender | of course |
| 17:45.39 | egassem | [TK]D-Fender: really |
| 17:45.44 | egassem | is that possoble |
| 17:46.07 | [TK]D-Fender | <[TK]D-Fender> of course <- |
| 17:46.25 | egassem | [TK]D-Fender: you talking to me |
| 17:46.35 | lorsungcu | you talking to ME? |
| 17:46.48 | egassem | now Im lost |
| 17:46.57 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 17:47.17 | lorsungcu | i thought we were doing the scene from taxi driver |
| 17:47.28 | darkunderlord_ | lorsungcu, lol I got it |
| 17:47.32 | lorsungcu | thx |
| 17:47.38 | lorsungcu | ;-) |
| 17:48.25 | *** join/#asterisk juvenal (~juvenal@177.197.88.89) |
| 17:48.42 | egassem | looool |
| 17:49.01 | egassem | you talking is somthing robert deniro said |
| 17:50.24 | egassem | so any one can help me |
| 17:50.33 | lorsungcu | egassem: you were answered |
| 17:50.36 | darkunderlord_ | they said you can use it |
| 17:50.54 | darkunderlord_ | should be a dahdi setting for each span of ports I'd think |
| 17:51.11 | egassem | should i go with libss7 or chan_dahdi |
| 17:51.17 | egassem | which is better in this case |
| 17:52.20 | egassem | one sec |
| 17:52.22 | egassem | brb |
| 17:52.24 | *** part/#asterisk egassem (86237344@gateway/web/freenode/ip.134.35.115.68) |
| 17:52.47 | *** join/#asterisk egassem (~egassem@134.35.61.92) |
| 17:52.58 | egassem | hi |
| 17:53.00 | egassem | im back |
| 17:53.21 | [TK]D-Fender | egassem> [TK]D-Fender: you talking to me <- YES |
| 17:53.24 | [TK]D-Fender | I answered twice |
| 17:53.30 | [TK]D-Fender | I answer RIGHT AFTER your question |
| 17:53.40 | egassem | that is possible |
| 17:53.45 | [TK]D-Fender | and then your SECOND "is that possible" |
| 17:53.46 | egassem | is there a guide for this |
| 17:53.52 | [TK]D-Fender | <[TK]D-Fender> <[TK]D-Fender> of course <- |
| 17:53.55 | egassem | sorry i was confused |
| 17:53.57 | [TK]D-Fender | YES IT'S POSSIBLE |
| 17:54.02 | [TK]D-Fender | what is hard to understand? |
| 17:54.32 | egassem | im not used to get an answer so quick so i got shocked |
| 17:54.34 | [TK]D-Fender | egassem> is there a guide for this <- thre is no magic. Each set of ports is configured jsut like any other |
| 17:54.35 | egassem | ok |
| 17:54.50 | [TK]D-Fender | look at the sample settings for each and set them according to the signalling you require |
| 17:55.03 | [TK]D-Fender | There are SS7 sample configs. There are E1 PRI sample configs. |
| 17:55.11 | [TK]D-Fender | There is nothing magical about having BOTH |
| 17:55.18 | egassem | do you recommed using chan_ss7 or libpri |
| 17:55.52 | [TK]D-Fender | if libpri does the job then that's 1 less piece required |
| 17:55.59 | [TK]D-Fender | I've never done SS7 personally |
| 17:56.14 | egassem | sorry i meant libss7 or chan_ss7 |
| 17:56.54 | egassem | ok i'll read more about it |
| 18:10.04 | *** join/#asterisk juvenal (~juvenal@177.197.88.89) |
| 18:16.53 | *** join/#asterisk juvenal (~juvenal@177.197.88.89) |
| 18:29.01 | igcewieling | does anyone know why I might be getting this error? I thought I was out of licenses, but I doubled the license count and it didn't seem to help. [2017-07-24 14:28:04] WARNING[18744][C-000034c3]: translate.c:433 ast_translator_build_path: No translator path from g729 to slin |
| 18:35.49 | lorsungcu | i think asterisk is mocking you for using g729 |
| 18:35.59 | *** join/#asterisk miralin1 (~Thunderbi@195.209.246.194) |
| 18:37.56 | igcewieling | Sometimes I wonder why I bother asking. |
| 18:38.24 | lorsungcu | same |
| 18:42.16 | *** join/#asterisk miralin (~Thunderbi@91.237.94.1) |
| 18:46.40 | drmessano | igcewieling: Sounds like you can't transcode |
| 18:47.16 | drmessano | Regardless of "License blah blah" |
| 18:47.29 | drmessano | Which asterisk version? |
| 18:47.33 | drmessano | Specifically |
| 18:47.48 | igcewieling | don't bother. you'll tell me to upgrade. |
| 18:48.00 | drmessano | Which asterisk version? |
| 18:48.02 | drmessano | Specifically |
| 18:48.04 | igcewieling | and I'm tired to justifying my decisions to the channel. |
| 18:48.15 | drmessano | Why dont you just tell me |
| 18:48.20 | drmessano | Maybe you have the wrong module |
| 18:48.23 | igcewieling | Asterisk 11.21.2 |
| 18:48.27 | drmessano | Okay |
| 18:48.46 | RovingWriter | if _# matches #, does _* match * ? |
| 18:48.47 | drmessano | Have you updated recently? |
| 18:48.59 | drmessano | When did you go to 11.21.2 |
| 18:49.03 | *** join/#asterisk miralin (~Thunderbi@195.209.246.194) |
| 18:49.08 | drmessano | What were you on before? |
| 18:49.32 | drmessano | You know they've updated the g729 modules |
| 18:50.24 | igcewieling | Feb 1 09:49 /usr/sbin/asterisk |
| 18:50.26 | drmessano | You should be running the codec_g729.so from 9/7/2016 |
| 18:50.41 | drmessano | You should be running the codec_g729a.so from 9/7/2016 |
| 18:51.53 | igcewieling | the timestamp on my codec_g29a is the same as my install time so I don't know the date for it. |
| 18:52.09 | drmessano | Was the install on feb 1/ |
| 18:52.11 | drmessano | Was the install on feb 1? |
| 18:52.52 | igcewieling | I don't know, but since the timestamp on asterisk is Feb 1, then I assume the last update was at that time. |
| 18:53.14 | drmessano | Has transcoding ever worked? |
| 18:53.36 | igcewieling | yes. |
| 18:53.42 | igcewieling | It works most mornings. |
| 18:53.58 | *** join/#asterisk miralin (~Thunderbi@91.237.94.1) |
| 18:54.29 | drmessano | 'g729 show licenses' is accurate... and 'core show translation' look fine? |
| 18:55.23 | igcewieling | core show translations shows it transcodes g729 |
| 18:55.43 | drmessano | 'g729 show licenses' is accurate... ? |
| 18:55.50 | drmessano | Two questions there |
| 18:56.58 | igcewieling | I don't know if the show licenses is correct. The TOTAL number if licenses is correct. I don't know if the inuse cound it. |
| 18:57.02 | igcewieling | count is. |
| 18:57.46 | drmessano | When did this start? Today? |
| 18:58.19 | igcewieling | About a week ago. I've been reducing the need for licenses and on Friday I added more licenses. |
| 18:58.42 | drmessano | Did you restart the instance? |
| 18:58.58 | igcewieling | yes. twice last week. |
| 18:59.24 | igcewieling | I've not today since the customer gets upset when we droop 20ish calls. |
| 19:00.11 | Samot | core show translation paths g729 |
| 19:00.19 | Samot | What's that show? ^^^ |
| 19:00.56 | igcewieling | https://pastebin.com/7wfD14Wf |
| 19:01.56 | Samot | Is it happening all the time or just when a certain amount of calls hit? |
| 19:02.40 | igcewieling | It works most mornings, noonish things fall apart. I assume it is related to call volume. |
| 19:02.57 | drmessano | How many licenses are on the server? |
| 19:03.48 | *** join/#asterisk shanth (62b67ee2@gateway/web/cgi-irc/kiwiirc.com/ip.98.182.126.226) |
| 19:04.28 | igcewieling | "0/38 encoders/decoders of 40 licensed channels are currently in use" with 19 active channels |
| 19:04.47 | shanth | ;exten => s,n,Set(CALLED_NUMBER=${HEADER_TO:5:10}) |
| 19:04.47 | shanth | exten => s,n,Set(CALLED_NUMBER=${HEADER_TO:5:10}) what does the 5:10 part mean? |
| 19:04.50 | shanth | whoops |
| 19:04.55 | Samot | How many existed before you added more on Friday? |
| 19:04.59 | Samot | You doubled it? |
| 19:05.07 | igcewieling | it has 20 before fridaty |
| 19:05.27 | Samot | And it seems to be about when they creep up to 20 calls it shits the bed? |
| 19:05.48 | Samot | Because it's sounding like something isn't honoring your new limit. |
| 19:06.01 | igcewieling | Samot: yes, both before AND after the extra licenses. |
| 19:06.06 | Samot | And is still stuck at thinking there is only 20 |
| 19:07.07 | igcewieling | I don't know. It says I have 40 licenses. |
| 19:07.32 | Samot | Well that's my theory. |
| 19:07.40 | Samot | It shows you have 40 but acting like you have 20 |
| 19:07.59 | Samot | You are the only one that can prove or disprove that theory but it's a jumping point. |
| 19:08.22 | shanth | asterisk project just fell in my lap. i have no idea what im doing lol |
| 19:08.45 | shanth | what kind of rope should i hang myself with? |
| 19:09.13 | igcewieling | shanth: your best option is to find another employer. |
| 19:09.32 | shanth | lol |
| 19:10.27 | shanth | just trying to go line by line and figure out what stuff does, very confusing |
| 19:11.38 | igcewieling | shanth: read the Asterisk book, read "core show channels" and "cores show functions" |
| 19:12.31 | shanth | thanks |
| 19:13.59 | shanth | do you mean this http://the-asterisk-book.com igcewieling? |
| 19:14.25 | shanth | or asterisk the definitive guide? |
| 19:15.35 | igcewieling | There is nothing for V12+ http://www.asteriskdocs.org/ < 1.8 and above |
| 19:17.54 | shanth | im running asterisk 11.25.1 |
| 19:18.10 | shanth | yay |
| 19:30.48 | RovingWriter | I have a queue, where I have periodic-announce=/path/to/my/file, and periodic-announce-frequency=10, but regardless, the queue spends about 60 seconds, then just uses the generic "all agents are busy" recording of Alice... any ideas why? I put those options within my queue's direct context, not in the general |
| 19:31.19 | RovingWriter | its like it just doesn't recognize those options. version is 1.8.11-1 |
| 19:32.27 | RovingWriter | nevermind. |
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| 19:33.38 | [TK]D-Fender | <shanth> exten => s,n,Set(CALLED_NUMBER=${HEADER_TO:5:10}) what does the 5:10 part mean? <- trim 1st 5, take 20 long after that |
| 19:33.44 | [TK]D-Fender | 10 rather |
| 19:36.05 | shanth | 'trim the first 5 and take 10 long after that' what do you mean when you say 10 long? |
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| 19:37.47 | igcewieling | 10 characters in length |
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| 19:40.44 | [TK]D-Fender | if it's 30 long then you'll get 10 chars starting at the 6th char |
| 19:40.59 | shanth | great thanks [TK]D-Fender |
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| 19:55.43 | shanth | im currently working on asterisk with a test envrionement that has a self signed ssl cert. is there a way to globally set the curlopt to ignore or bypass the cert? i was trying to do it per line like ;exten => s,n,CURLOPT(ssl_verifypeer)=0 |
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| 20:11.11 | klow | Trying to track down how I am crashing asterisk 14.3.0 , anyone mind having a look? https://pastebin.com/nycFG25n it is happening when I apply config in freepbx UI (manually installed/compiled) |
| 20:11.32 | igcewieling | Here is a nice complicated one for you to parse. lol. CELGenUserEvent(SM_${IF($["${MASTER_CHANNEL(sm_call[${call_index}][dialstatus])}" == "ANSWER"]?HANGUP:${MASTER_CHANNEL(sm_call[${call_index}][dialstatus])})},tag='ast' hangup_source='dest' dialstatus='${MASTER_CHANNEL(sm_call[${call_index}][dialstatus])}' ... hangup_timestamp='${STRFTIME(,,%s.%3q)}'); |
| 20:21.51 | igcewieling | klow: anyone who can help you will tell you to first upgrade to the latest Asterisk 14.x |
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| 20:28.33 | klow | fair enough i will compile 14.6 then ;) |
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| 21:26.30 | shanth | my curl call is failing because of the self signed ssl cert http://dpaste.com/3WKJSHW - how can i make it bypass this? |
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