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| 06:24.06 | _8bits | Is there any other way to make "Dial another peer if first one is offline" logic besides queue? |
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| 12:34.56 | gavimobile | havent used my pbx in a while and not sure why but my outgoing google calls arent working. im using simon gsgw's paid $5 gateway. can someone give me a hand please? https://pastebin.com/1ygxTA4k |
| 12:37.12 | gavimobile | i may have not added a 1 before the number. here is the debug with the 1 https://pastebin.com/7uuFWxyP |
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| 13:21.42 | Samot | gavimobile: Google Voice is dead. Google announced it's death in February. It died June 26th. |
| 13:23.43 | Samot | Need to find an alternative. |
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| 13:27.16 | gavimobile | Samot: June 26th 2017? |
| 13:27.31 | Samot | Uhm. |
| 13:27.35 | Samot | Yes. |
| 13:27.55 | gavimobile | Samot: also, what features are not working, because incoming IS and has been working |
| 13:28.08 | Samot | XMPP connectors. |
| 13:28.18 | Samot | The thing that makes connecting to them possible. |
| 13:28.20 | gavimobile | Samot: i said i was using gvgw |
| 13:28.25 | gavimobile | that doesnt use xmpp |
| 13:28.29 | Samot | Uhm. |
| 13:28.31 | Samot | Come on dude. |
| 13:28.43 | gavimobile | ? |
| 13:28.47 | Samot | There has been one way to connect to Google Voice |
| 13:28.50 | Samot | XMPP |
| 13:28.58 | Samot | That service is someone running that module FOR YOU |
| 13:29.09 | Samot | And acting as the SIP to XMPP gateway |
| 13:29.40 | gavimobile | Samot: i see. so right now gv outgoing isnt working for anyone? |
| 13:29.57 | Samot | Probably not. They been slowing letting it die. |
| 13:30.10 | Samot | XMPP node dies, they don't fix it |
| 13:30.13 | gavimobile | Samot: that i know, which is why i left motiff and moved to gvgw |
| 13:30.26 | Samot | And as of June 26th they are official dead. |
| 13:30.37 | gavimobile | thats pretty recent. crazy |
| 13:30.37 | Samot | So there may be some lingering ones that might work but it is done. |
| 13:30.56 | gavimobile | Samot: and does incoming use xmpp as wel? |
| 13:30.57 | Samot | They've been announcing the death of GV for over a year |
| 13:31.01 | Samot | Yes. |
| 13:31.15 | Samot | They finally put a date to the death in February. |
| 13:31.23 | gavimobile | Samot: ohhh, my incoming workings because gvoice is forwarding to my free ipcomm did number |
| 13:31.48 | gavimobile | so that wouldnt use xmpp, correct? |
| 13:31.49 | Samot | Right, Google Talk/Hangouts are still working |
| 13:31.56 | Samot | So those run Google Voce. |
| 13:31.59 | gavimobile | Samot: thanks... |
| 13:32.03 | Samot | Internally Google Voice works. |
| 13:32.07 | gavimobile | do you have a official link from google |
| 13:32.16 | Samot | As long as it is on the gmail network and no third party connectors. |
| 13:32.32 | Samot | I don't have the link from the announcement sent almost 7 months |
| 13:32.50 | gavimobile | also, does the asterisk community have any leads? |
| 13:33.13 | gavimobile | maybe i should invest in a magic jack. i have a pcie analog card |
| 13:40.47 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:42.43 | igcewieling | gavimobile: get an account at an ITSP and get on with your life. |
| 13:43.18 | igcewieling | I use Vitelty, but there are dozens of them. |
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| 13:59.01 | gavimobile | igcewieling: i live over seas. i dont make much calls to the us. |
| 13:59.24 | gavimobile | i do need to call my parents every so often which is why i like google voice. |
| 13:59.39 | gavimobile | what about whatsapp, has anyone been able to create a sip trunk with whatsapp |
| 14:00.03 | gavimobile | gavimobile: nevermind, scratch that out. |
| 14:00.26 | Samot | Uhm. |
| 14:00.39 | Samot | Why don't you just ship them an ATA that registers to Asterisk? |
| 14:00.49 | Samot | Then just call them via the local extension? |
| 14:01.00 | Samot | $0 call. |
| 14:01.24 | gavimobile | its extra clutter that my mom will not be happy about |
| 14:01.31 | gavimobile | she can always call on whatsup |
| 14:01.31 | Samot | OK. |
| 14:01.32 | Samot | Sure. |
| 14:01.44 | Samot | Well Whatsapp does not work with SIP |
| 14:02.12 | gavimobile | im thinking something along the lines of magic jack pro to a pcie card |
| 14:02.22 | Samot | Sure. |
| 14:02.44 | Samot | I mean if your mom doesn't have space for a 6"x6" device |
| 14:02.58 | Samot | Sure, spend way more time and money getting a free calling solution in place. |
| 14:03.06 | gavimobile | i have the card |
| 14:03.24 | Samot | OK. |
| 14:03.29 | Samot | So time to make it work |
| 14:03.34 | Samot | Money to pay for MagicJack |
| 14:03.47 | Samot | Like I said, spend way more time and money to get a free calling solution in place. |
| 14:04.01 | gavimobile | Samot: im open for suggestions |
| 14:04.13 | gavimobile | i do like the comfort to be able to call other american numbers |
| 14:04.19 | Samot | I just gave the best one for free overseas calls |
| 14:04.23 | Samot | OK. |
| 14:04.27 | Samot | Then you need a US service. |
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| 14:06.42 | gavimobile | is there a service like magic jack but for less? |
| 14:08.37 | *** join/#asterisk imcdona (~imcdona@64.122.164.5) |
| 14:09.39 | Samot | I have no idea. |
| 14:09.44 | Samot | I don't use those services. |
| 14:11.08 | [TK]D-Fender | How much is MagicJack charging and how much do you need? |
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| 14:29.14 | igcewieling | I pat about $2/month and $0.09/min, how much cheaper do you need?" |
| 14:29.27 | igcewieling | s/pat/pay |
| 14:35.50 | [TK]D-Fender | I think your /min is a bit off... |
| 14:36.07 | igcewieling | 9 cents per min |
| 14:36.20 | igcewieling | sorry, you are right |
| 14:36.27 | igcewieling | $0.009/min |
| 14:36.37 | [TK]D-Fender | #verizonmath |
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| 14:55.39 | libardi | #join /asterisk-dev |
| 14:55.59 | Martin` | <PROTECTED> |
| 14:56.31 | libardi | Sorry :) |
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| 15:31.47 | drmessano | wow |
| 15:31.51 | drmessano | 09:30:14 <gavimobile> Samot: that i know, which is why i left motiff and moved to gvgw |
| 15:32.12 | drmessano | So you're paying $5 a month to someone to basically do exactly what you were doing |
| 15:32.59 | drmessano | Nm.. one time fee |
| 15:33.04 | drmessano | But still |
| 15:33.11 | drmessano | They were using XMPP just the same lol |
| 15:33.27 | Samot | 9:28:58 AM <Samot> That service is someone running that module FOR YOU |
| 15:33.27 | Samot | 9:29:09 AM <Samot> And acting as the SIP to XMPP gateway |
| 15:33.40 | drmessano | Yep |
| 15:33.42 | Samot | But yeah, that was the actually reply to me pointing that out. |
| 15:34.26 | drmessano | I guess I didn't make enough money off GV while it lasted |
| 15:34.37 | drmessano | Good god.. All I had to was act as a faux gateway |
| 15:34.45 | drmessano | to do* |
| 15:35.00 | file | remembers when someone ran a call center off Google Voice |
| 15:36.12 | drmessano | I think the lack of information/misinformation that circulated during its lifespan was sad |
| 15:36.15 | drmessano | Such as |
| 15:36.29 | drmessano | "The XMPP stuff in Asterisk doesn't work, use my SIP gateway" |
| 15:36.37 | drmessano | Well, the OLD modules didnt work |
| 15:36.56 | drmessano | But hey, needed to start somewhere |
| 15:37.07 | drmessano | I never had any problems with the new ones |
| 15:37.22 | drmessano | But paying someone $6 to |
| 15:37.28 | drmessano | Oh god.. I just realized |
| 15:37.45 | drmessano | Paid $6 to act as a Google Proxy |
| 15:37.50 | drmessano | BA-DUMP-CHING |
| 15:38.27 | drmessano | But yeah |
| 15:38.48 | drmessano | "Obi-Hai has a backdoor into Google" |
| 15:39.15 | drmessano | "They can't talk about it.. NDA and all" |
| 15:39.41 | drmessano | Best and worst thing that ever happened to telephony |
| 15:51.18 | *** join/#asterisk mub (~jub@unaffiliated/mub) |
| 15:59.17 | RovingWriter | file, i too ran a call center off of GV |
| 15:59.39 | RovingWriter | for 3 months, then switched to flowroute once we were making $ haha |
| 16:00.02 | Samot | Yeah, I never would. |
| 16:01.14 | RovingWriter | i didn't have many options. funds were super tight hah |
| 16:01.22 | RovingWriter | but it worked for 3 months pretty well. |
| 16:03.38 | file | Google frowns upon it if the abuse logic detects it. |
| 16:04.27 | drudge` | i been using questblue or something for sip trunks, its not bad |
| 16:08.37 | Samot | I just would never put a call center on something as unstable as GV. |
| 16:08.44 | Samot | That has no ToS |
| 16:08.49 | Samot | No SLA |
| 16:08.59 | Samot | In fact, no guarantee that service would work. |
| 16:09.26 | drmessano | RovingWriter: How did you have a call center but no budget for the actual "call" part? |
| 16:09.32 | Samot | Yeah. |
| 16:09.38 | Samot | There is that. |
| 16:09.49 | drmessano | I hate when people say "no budget" |
| 16:09.53 | Samot | Well.. |
| 16:10.01 | Samot | It was like back in the day |
| 16:10.02 | drmessano | It's a lie, basically |
| 16:10.07 | Samot | When people would want free email |
| 16:10.18 | Samot | Then complain how they were losing $10,000 because the free service wasn't working |
| 16:10.58 | drmessano | If you don't have a budget for the "cost of doing business" stuff, you shouldn't be in business |
| 16:11.12 | Samot | Putting a call center on something like GV is like being a VoIP provider using GV. |
| 16:11.28 | Samot | People are paying you for expectations of service... |
| 16:11.32 | drmessano | There are a lot of things I would LIKE to have |
| 16:11.37 | drmessano | Improvements, even |
| 16:11.47 | drmessano | But that GV thing is just being cheap |
| 16:11.50 | Samot | Not that they are paying you for some free service that has not guarantees. |
| 16:11.53 | drmessano | Trying to rip off Google |
| 16:12.33 | drmessano | Like my coworker that doesn't want a 3rd AP at home |
| 16:12.41 | drmessano | but can't wee-fee all over |
| 16:12.49 | drmessano | Just got a $600 TV |
| 16:12.55 | drmessano | No budget for a 3rd AP |
| 16:13.07 | Samot | Not even the $30 for the new little mini? |
| 16:13.10 | drmessano | No, "doesn't want to buy it" |
| 16:13.24 | drmessano | That's single band |
| 16:13.30 | Samot | Oh well |
| 16:13.34 | *** join/#asterisk znf (~ibm86@toaster.linge-ma.ws) |
| 16:13.35 | Samot | 'Cuse me. |
| 16:13.50 | drmessano | Does 2.4 work in your place? |
| 16:14.00 | Samot | Yeah |
| 16:14.03 | Samot | But I have dual. |
| 16:14.05 | drmessano | It doesn't in his place |
| 16:14.06 | drmessano | At all |
| 16:14.08 | drmessano | Zero |
| 16:14.08 | Samot | OK |
| 16:14.10 | Samot | So yeah |
| 16:14.11 | Samot | OK |
| 16:14.18 | Samot | So $45 for the dual? |
| 16:14.18 | RovingWriter | drmessano, so the call center was 100% remote, people working at home by calling into the PBX and hotdesking in. |
| 16:14.33 | drmessano | RovingWriter: .... so? |
| 16:14.36 | Samot | RovingWriter: That has nothing to do with the calls getting to the PBX. |
| 16:14.44 | Samot | And the guarantee that the users can make those calls |
| 16:14.57 | Samot | You could have 50 agents signed in... |
| 16:14.58 | RovingWriter | Samot, it wasn't for business uses |
| 16:15.00 | drmessano | Samot: Well, he's not gonna pay $45.. but not much more after markup for setting it up |
| 16:15.05 | Samot | What? |
| 16:15.10 | drmessano | lol |
| 16:15.15 | Samot | You have a non-business use call center? |
| 16:15.18 | drmessano | A call center not for business uses? |
| 16:15.27 | RovingWriter | it was for a charity thing. |
| 16:15.32 | drmessano | LOL |
| 16:15.39 | drmessano | How is that not "business use"? |
| 16:16.02 | drmessano | Sorry, English is my first language |
| 16:16.26 | drmessano | But I think you're confusing non-profit and not-business |
| 16:16.44 | Samot | 11:59:39 AMÂ <RovingWriter>Â for 3 months, then switched to flowroute once we were making $ haha |
| 16:16.51 | drmessano | Oh |
| 16:16.53 | drmessano | I missed that |
| 16:16.57 | Samot | So it was non-business, charity |
| 16:17.03 | drmessano | But making money |
| 16:17.08 | RovingWriter | yes, after 3 months we had raised some $$ and switched to flowroute to make calls |
| 16:17.19 | RovingWriter | then after 6 more months had met goal, and closed it |
| 16:17.20 | drmessano | But I think you're confusing non-profit and not-business <--- |
| 16:17.38 | RovingWriter | perhaps, i am not properly using correct wording |
| 16:17.44 | Samot | Well.. |
| 16:17.52 | Samot | 100% of all those usage charges.. |
| 16:17.58 | Samot | For a Non-Profit are tax writeoffs |
| 16:18.11 | Samot | So going with a free service for a non-profit doesn't benefit anyone |
| 16:18.53 | RovingWriter | sure, if you have the $$ to do it differently, you do. I didn't. |
| 16:19.04 | Samot | $0.0098/minute |
| 16:19.09 | RovingWriter | wasn't even a lot of $$ needed, but still more than I could do alone at the time |
| 16:19.34 | Samot | Well.. |
| 16:19.38 | RovingWriter | i don't even know why we're arguing about this, lol |
| 16:19.44 | Samot | Well |
| 16:19.49 | Samot | At first this was "call centers on GV" |
| 16:19.54 | Samot | Which I said, I wouldn't do. |
| 16:20.02 | Samot | Then you expanded your story. |
| 16:20.13 | Samot | But all I'm saying is next time.. |
| 16:20.17 | Samot | There are benefits you missed out on. |
| 16:20.22 | Samot | Both you and the non-profit. |
| 16:20.33 | RovingWriter | yes, i agree, thats why i moved it as soon as we couled |
| 16:20.34 | RovingWriter | could |
| 16:20.40 | Samot | No. |
| 16:20.46 | Samot | NM. |
| 16:21.11 | RovingWriter | i know what you are saying. spend the $$ you didn't have to do it in the most ideal way possible to start |
| 16:22.56 | Samot | Actually, no. |
| 16:23.05 | Samot | See non-profits still function like a normal business. |
| 16:23.16 | Samot | They cannot do it / make/ have profits from it. |
| 16:23.42 | Samot | But they can have "administrative costs" i.e. the cost of doing business. |
| 16:24.06 | Samot | Utilities (like phone service), staff payroll... |
| 16:24.17 | Samot | All those are write-offs |
| 16:24.30 | Samot | You donating phone service would have benefited you |
| 16:24.38 | Samot | It would allow them to mark that service as a donation. |
| 16:24.58 | Samot | All around it benefits doing it right way. |
| 16:26.00 | *** join/#asterisk karelk (~karel@31.10.154.117) |
| 16:26.30 | RovingWriter | here's, i'll expand a bit on the situation so you can poke at it a bit more. maybe it will clarify it a bit. I have a son who is now 3 years old. when he was born, he had a stroke as being born. then the seizured came. every few days.... broke me financially... like, almost homeless status. I have to stay home with him 24/7. so I picked up programming in order to be able to make $$ while at |
| 16:26.30 | RovingWriter | home. because his seizures happen every 2 weeks now, and there is no real way to detect them or when they will happen, i reached out to a company called 4 paws 4 humanity. they train service dogs to detect things like seizures, and all sorts of stuff.... they quoted me $15k for the dog.... so I, while juggling all the medical bills, copays, rent, food, oxygen machines, rescue meds, etc... |
| 16:26.30 | RovingWriter | setup a cloud based * install on digitalocean (i know, I shouldn't run it in the cloud either), and linked it to GV... then friends and family linked their phones to it via sip clients, and we called animal shelters, and asked for small donations in order to raise the $$ to buy the dog. |
| 16:26.55 | Samot | There's nothing wrong with it being in the cloud |
| 16:27.03 | RovingWriter | so, overall, nothing was technically done right in the situation. |
| 16:27.16 | Samot | Then this wasn't a non-profit. |
| 16:27.27 | Samot | This was "gofundme" style. |
| 16:27.45 | Samot | So not a call center. |
| 16:27.50 | Samot | Not a non-profit. |
| 16:27.59 | Samot | So OK, GV not a bad choice. |
| 16:28.24 | RovingWriter | i guess i consider it a call center a bit because there were around 15 people calling out at some times |
| 16:28.57 | Samot | I guess. |
| 16:29.09 | RovingWriter | super loose definition i suppose. |
| 16:29.20 | RovingWriter | anyhow. |
| 16:29.25 | Samot | That's like all the parents in a girl scout troop using Asterisk to make calls for selling cookies |
| 16:29.38 | Samot | Than the parents using their cell phones. |
| 16:31.57 | Samot | Sorry, I work in Telecom. So saying you ran a "call center on GV" is triggerish for me. |
| 16:32.22 | Samot | Saying that you used GV and ASterisk so you and 15 friends could raise $$ for a dog... |
| 16:32.30 | Samot | That makes complete sense. |
| 16:33.07 | RovingWriter | i suppose, yeah. its like me saying I'm a mechanic because i swapped a clutch in my car |
| 16:33.08 | RovingWriter | i can see that |
| 16:33.22 | Samot | Well |
| 16:33.29 | Samot | If you walked into a room of mechanics... |
| 16:33.38 | Samot | Professionals |
| 16:33.41 | Samot | Not backyard |
| 16:33.46 | RovingWriter | yeah |
| 16:33.48 | Samot | And started talking about things like that |
| 16:34.11 | Samot | Without the qualifier of "backyard", yes you would probably have the same reaction by someone. |
| 16:34.43 | RovingWriter | yeah, i can see that |
| 16:35.26 | Samot | Because in the bad of their heads they are thinking "Why is this pro talking like a hobbyist?" |
| 16:36.14 | Samot | Believe me, professionals are probably the biggest hobbyists about shit... |
| 16:37.23 | RovingWriter | yeah |
| 16:37.34 | Samot | I've used Asterisk for home stuff in various ways over the years. |
| 16:37.41 | Samot | Nothing I would ever do professionally.. |
| 16:37.42 | RovingWriter | I have built a legit call center since then though too ;) |
| 16:37.59 | Samot | And would you use GV for a legit call center? |
| 16:38.03 | RovingWriter | nope |
| 16:38.06 | Samot | Right |
| 16:38.08 | Samot | See our point |
| 16:38.17 | RovingWriter | yes of course |
| 16:38.30 | RovingWriter | i wasn't defending it as a good choice |
| 16:38.37 | RovingWriter | at least i hope that wasn't coming off |
| 16:39.00 | Samot | Well for a legit call center no... |
| 16:39.05 | Samot | But for what you did |
| 16:39.08 | Samot | Yeah, it was fine. |
| 16:40.02 | RovingWriter | raised $21k in 9 months. 3 of which were on GV. I should go back through CDR and see if we did better after switching, haha |
| 16:40.17 | Samot | Well it was outbound |
| 16:40.22 | RovingWriter | yeah it was |
| 16:40.27 | Samot | So.. |
| 16:40.40 | Samot | Doing better after switching would mean you just got your sales pitch down. |
| 16:40.53 | Samot | The difference is inbound. |
| 16:40.56 | RovingWriter | well, there was a lot of "hello? i can't hear you" |
| 16:41.01 | RovingWriter | before switching |
| 16:41.04 | Samot | OK. |
| 16:41.17 | RovingWriter | like... 1 in 20 calls probably |
| 16:41.25 | Samot | Well |
| 16:41.28 | Samot | Everyone was remote |
| 16:41.32 | RovingWriter | yeah |
| 16:41.36 | Samot | NAT and god knows what crap routers |
| 16:41.36 | *** join/#asterisk ghost75 (~quassel@p200300C783C29A00505400FFFE6A8831.dip0.t-ipconnect.de) |
| 16:41.40 | RovingWriter | lotsa variables |
| 16:42.09 | Samot | But inbound call centers are the more common place. |
| 16:42.09 | ghost75 | does * support 3way authentication with sip? |
| 16:42.36 | Samot | ghost75: If you mean multiple contacts/AORs then yes with the Chan_PJSIP driver. |
| 16:42.55 | ghost75 | 2 passwords for 1 sip account |
| 16:43.00 | Samot | No |
| 16:43.06 | Samot | Why would you need that? |
| 16:43.20 | ghost75 | provider needs it :< |
| 16:43.38 | Samot | One for inbound and one for outbound? |
| 16:43.50 | ghost75 | no, for registration |
| 16:43.50 | Samot | That would be the only way they would "need it" |
| 16:44.06 | Samot | There is no "two password" registration method |
| 16:44.29 | ghost75 | i found only 3cx has it |
| 16:45.15 | Samot | That's because it's not a SIP standard |
| 16:45.23 | Samot | And they are a proprietary system. |
| 16:45.47 | Samot | And it only works with their softphone client. |
| 16:46.21 | ghost75 | telekom sip account needs that in order to register |
| 16:46.34 | Samot | What do they need? |
| 16:46.34 | ghost75 | otherwise only ip based |
| 16:46.38 | Samot | What did they send you? |
| 16:46.48 | Samot | Because there is no two password auth mech for SIP |
| 16:46.50 | ghost75 | they send nothing, this is germany :> |
| 16:48.14 | ghost75 | is telekom also offering sip in US ? |
| 16:48.58 | Samot | I have no clue |
| 16:49.42 | Samot | But SIP auth is based on the user:secret:domain |
| 16:49.53 | Samot | Not user:secret1:secret2:domain |
| 16:50.28 | *** join/#asterisk salviadud (~ralfalfa@189-211-190-134.static.axtel.net) |
| 16:50.38 | ghost75 | maybe its some "extra" field |
| 16:50.59 | Samot | Without seeing their creds/settings they sent for you to program your PBX with.. |
| 16:51.01 | Samot | No idea. |
| 16:51.04 | Samot | It could be. |
| 16:51.11 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
| 16:51.13 | Samot | Could be they want the fromuser= set to something different |
| 16:51.19 | ghost75 | basically the second pw is just email address |
| 16:52.31 | ghost75 | auth-id and user-id maybe? |
| 16:52.40 | Samot | OK |
| 16:52.50 | Samot | So yes, that is completely possible. |
| 16:53.07 | ghost75 | http://help.xnet.co.nz/questions/121/3CX+v10.0+Setup+instructions cause here they state this |
| 16:53.08 | Samot | But still only one secret- |
| 16:53.43 | Samot | Why does the 3CX configuration have to be involved? |
| 16:54.38 | Samot | 3Way Authentication <-- This is a Non-thing. |
| 16:54.52 | Samot | That just means "User AuthID for authorization" |
| 16:55.15 | ghost75 | its misleading |
| 16:55.19 | Samot | Well |
| 16:55.34 | Samot | First, you are looking at how to setup a completely different PBX system with their service. |
| 16:55.36 | Samot | So yes. |
| 16:55.50 | Samot | Instructions and setting names in 3CX will not match Asterisk. |
| 16:56.05 | Samot | Nor would the probably match another PBX systems |
| 16:56.11 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 16:56.15 | Samot | You need to find the Asterisk instructions for their service. |
| 16:56.38 | Samot | And not try to use a different PBXes instructions |
| 16:56.40 | ghost75 | they dont care about other services than their own :> |
| 16:56.49 | Samot | Who? |
| 16:56.55 | Samot | The provider? |
| 16:56.55 | ghost75 | german telekom |
| 16:57.22 | Samot | A provider should/could have instructions on how to connect many different PBX systems to their service. |
| 16:57.43 | Samot | That is a document explaining how to connect a 3CX PBX to their service. |
| 16:57.50 | Samot | Now that's great if you have a 3CX |
| 16:57.57 | Samot | And also one that is 5 years old |
| 16:58.06 | Samot | Because that's how old v10 is. |
| 16:58.26 | Samot | It's not even a supported version of that PBX. |
| 16:59.57 | Samot | http://www.rotherland.de/en/voip.html |
| 17:00.17 | Samot | This is someone who apparently has made the service work on Asterisk using Chan_PJSIP, no instructions for Chan_SIP |
| 17:00.31 | Samot | But you know, that was like the second link in my google search |
| 17:00.45 | Samot | Sorry first |
| 17:01.53 | Samot | This was the second, https://www.reddit.com/r/germany/comments/48slr6/telekom_voip_setup_question/ |
| 17:02.01 | Samot | Which has actual configs for Chan_SIP. |
| 17:02.39 | Samot | puts up the LMGTFY "out to lunch" sign. |
| 17:03.25 | ghost75 | authuser=$YOUR_T-ONLINE_MAIL_ADDRESS |
| 17:03.51 | ghost75 | just asked for that 3way thing :) |
| 17:04.09 | ghost75 | so now its clear that its just authuser |
| 17:04.39 | Samot | Right |
| 17:05.07 | Samot | But you were trying to figure how what 3CX setting equaled what in Asterisk. |
| 17:05.27 | ghost75 | right |
| 17:05.38 | Samot | telekom germany asterisk setup <-- In Google, those were the top two links. |
| 17:07.19 | RovingWriter | Samot, here ya go, only pics I have: http://imgur.com/a/QuEye |
| 17:07.20 | RovingWriter | hehe |
| 17:15.15 | *** join/#asterisk miralin (~Thunderbi@91.237.94.1) |
| 17:18.23 | tcpdump | GM everyone. |
| 17:20.17 | tcpdump | So - question: What exactly is WebRTC in comparison to SIP? Is it just a SIP client, or is it something a bit different? |
| 17:20.45 | Samot | Yes. |
| 17:20.57 | Samot | It uses Web Sockets to make the SIP connection. |
| 17:21.10 | Samot | Either the WSS or WS transports. |
| 17:22.10 | Samot | It is to facilitate browser based SIP clients. |
| 17:24.07 | *** join/#asterisk Gullibaer (~Signor_Ro@ip-109-90-16-90.hsi11.unitymediagroup.de) |
| 17:24.20 | Samot | I can't remember but file will probably know. Is DTLS needed when doing just WS or is it only for when you use WSS? |
| 17:24.40 | Samot | I'm shaking on that since I've only ever done it via WSS and DTLS was needed. |
| 17:24.45 | file | WebRTC requires DTLS. |
| 17:24.50 | Samot | OK |
| 17:25.01 | Samot | So you need to get a cert for it as well |
| 17:25.24 | file | WebRTC itself is two parts: a set of standards (ICE, STUN, TURN, DTLS, SDP, some other things) and an interface the browser provides via Javascript to use things |
| 17:26.12 | Samot | With the right client and configuration, it's pretty decent. |
| 17:28.53 | *** join/#asterisk Maliuta_ (maliutamat@gateway/shell/matrix.org/x-zdlsjcecipdafipo) |
| 17:30.06 | *** join/#asterisk bananapie (~david@modemcable162.109-37-24.static.videotron.ca) |
| 17:30.53 | bananapie | Is there a way from asterisk manager to find a channel based on a variable value without having to load all the channels and do a get variable on each ? Like SQL : SELECT ChannelName FROM Asterisk WHERE SomeVariable = 'SomeValue' ? |
| 17:31.32 | Samot | Not that I am aware of |
| 17:31.42 | Samot | Since those variables are attached to their channels |
| 17:31.52 | igcewieling | bananapie: Manager, not an AGI script? |
| 17:32.02 | bananapie | Manager, not an AGI script. |
| 17:32.26 | Samot | Manager needs to know the channel is it managing |
| 17:32.30 | bananapie | I am originating a call via manager, and I want to wait for the call to end. The call uses a local channel, so when answered, the local channel is optimised out. |
| 17:32.33 | Samot | With the channel id |
| 17:32.42 | igcewieling | It would be trivial when using AGI. |
| 17:32.46 | bananapie | I don't want to use events ( too much volume ) and /n messes up transfers. |
| 17:32.47 | bananapie | yes. |
| 17:32.52 | Samot | OK.. |
| 17:32.52 | bananapie | it would be. |
| 17:33.27 | Samot | So Leg A of the Local channel makes the call... |
| 17:33.51 | Samot | When the destination answers, Leg A drops and switches to Leg B |
| 17:33.54 | bananapie | I originate to Channel Local/someextension@somecontext, and that local channel calls a SIP peer |
| 17:33.59 | Samot | Which is bridged to the other channel. |
| 17:34.02 | Samot | Right |
| 17:34.02 | igcewieling | bananapie: You could save the info in a database table when the channel is created, then search it as you normally would. |
| 17:34.08 | Samot | Local is a two leg channel |
| 17:34.12 | Samot | It is outbound first |
| 17:34.19 | bananapie | yea, I did that. exten => answer,n,Set(DB(OriginateChannel/${MY_CALL_ID})=${CHANNEL}) |
| 17:34.22 | bananapie | but something went wrong. |
| 17:34.27 | Samot | And when the destination answers, it becomes inbound. |
| 17:34.40 | bananapie | ok. I'll keep playing with the database. I think that the solution is there, just need some tinkering. |
| 17:34.41 | bananapie | thanks :) |
| 17:34.50 | Samot | Like the destination was the one to originate the call. |
| 17:34.51 | tcpdump | So, I think you inadvertantly answered my second question file . I was under the impression that a STUN/TURN server together was a ICE (much like Elastic Search, Kibana, Logstash is ELK) but it seems that ICE is a different protocol all together? |
| 17:35.37 | file | it's a separate thing which uses STUN and TURN underneath |
| 17:35.53 | tcpdump | I see. |
| 17:36.31 | *** join/#asterisk bananapie (~david@modemcable162.109-37-24.static.videotron.ca) |
| 17:48.28 | *** join/#asterisk joshelson (~joshelson@64.55.128.243) |
| 17:49.14 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 17:49.49 | *** join/#asterisk znf (~ibm86@toaster.linge-ma.ws) |
| 17:53.44 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 18:03.11 | shanth | having a hard time figuring out how to make the timeout extension work http://dpaste.com/07S8TYY |
| 18:03.35 | shanth | shouldnt this jump to timeout after 5 seconds? |
| 18:04.22 | shanth | also tried with WaitExten with no luck |
| 18:04.34 | Samot | same => n,Wait(100) <-- that's how long to wait |
| 18:04.37 | Samot | In seconds. |
| 18:04.51 | Samot | Setting a variable TIMEOUT doesn't do anything. |
| 18:05.18 | Samot | n,WaitExten(5) <-- Wait for input for 5 seconds |
| 18:05.26 | Samot | That will trigger the t extension. |
| 18:05.55 | shanth | let me try that |
| 18:10.24 | shanth | Samot if i call and press nothing, it doesnt jump to time out, it only jumps to timeout if i enter a partial entry and wait |
| 18:10.28 | shanth | is that expected? |
| 18:11.58 | Samot | WaitExten(5) is just a response in 5 seconds. |
| 18:12.02 | Samot | Regardless of length |
| 18:12.26 | Samot | At that point that i extension comes into place. |
| 18:12.44 | Samot | If what they sent doesn't match |
| 18:12.54 | shanth | if no response is given after WaitExten is the proper thing to do proceed tdo the next step or jump to timeout? http://dpaste.com/1S35NYV |
| 18:13.26 | Samot | WaitExten is "Wait X seconds for DTMF input" |
| 18:13.42 | Samot | If not DTMF input and 5 seconds passes, it's a timeout |
| 18:13.58 | Samot | If there is DTMF input and there is nothing to match, invald |
| 18:14.01 | Samot | If there is DTMF input and there is nothing to match, invalid |
| 18:14.05 | shanth | it's not jumping to timeout, it's going straight to the next step which is hangup |
| 18:14.13 | igcewieling | Most of the time I use Read |
| 18:14.39 | shanth | it should be going to timeout, playing goodbye and hanging up, it's just going to hangup which is under WaitExten, why is that? |
| 18:14.54 | Samot | Show it |
| 18:14.59 | shanth | http://dpaste.com/1S35NYV |
| 18:14.59 | Samot | asterisk -rvvvvvvvvvv |
| 18:15.01 | Samot | ~pb |
| 18:15.01 | infobot | pastebin is, like, 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. |
| 18:15.06 | Samot | Not the dialplan. |
| 18:15.09 | Samot | The actual call |
| 18:15.12 | Samot | The attempt |
| 18:15.12 | shanth | i will paste it |
| 18:16.25 | *** join/#asterisk karelk (~karel@31.10.154.117) |
| 18:16.40 | shanth | http://dpaste.com/2H891CB |
| 18:17.33 | shanth | are you sure that you aren't supposed to set timeout variables? i mean they dont seem to do anything but idk |
| 18:18.40 | igcewieling | shanth: you've read the output of "core show function TIMEOUT"? |
| 18:18.55 | Samot | shanth: Is this dialplan define in a context? |
| 18:19.03 | Samot | Or is just in the dialplan? |
| 18:19.13 | shanth | it's in a context |
| 18:21.20 | igcewieling | of course you need to set a timeout. |
| 18:21.48 | igcewieling | here is a sample IVR, but for an older version of Asterisk. https://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu |
| 18:22.12 | Samot | No. |
| 18:22.13 | Samot | FFS |
| 18:22.28 | Samot | WaitExten(5) <-- Wait 5 seconds before TIMING OUT |
| 18:22.36 | shanth | that's not what it's doing though samot |
| 18:22.41 | Samot | I understand that |
| 18:22.46 | Samot | I think it's the dialplan |
| 18:22.54 | Samot | Change same=> to exten => |
| 18:23.25 | Samot | The only time the global TIMEOUT is used is if you just do WaitExten() |
| 18:23.34 | Samot | With no timeout option. |
| 18:24.33 | igcewieling | There are three timeouts which can be set (four if you count WaitExten) |
| 18:25.08 | Samot | <PROTECTED> |
| 18:25.13 | Samot | That is your issue |
| 18:25.27 | Samot | It will never follow through to the t if there is a hangup before it |
| 18:25.35 | Samot | Or the i |
| 18:26.14 | igcewieling | more sample IVRs to use as samples: http://www.asteriskguru.com/tutorials/ivr.html |
| 18:26.18 | Samot | https://www.irccloud.com/pastebin/ZBzLPoFY/ |
| 18:26.28 | Samot | That's just a snippet |
| 18:26.45 | Samot | But that code handles 1800 call batches and works just fine. |
| 18:26.57 | Samot | by 1800, I mean 1,800 calls |
| 18:27.02 | Samot | Not 1-800 |
| 18:27.05 | shanth | let me try yours |
| 18:27.12 | Samot | Well |
| 18:27.13 | Samot | No. |
| 18:27.22 | Samot | Drop the hangup |
| 18:27.24 | shanth | mine wont dial now, hmm |
| 18:27.28 | Samot | Show it |
| 18:27.32 | Samot | Show what you changed |
| 18:30.36 | shanth | just redoing it samot |
| 18:31.29 | Samot | https://www.irccloud.com/pastebin/nhdUPwqg/ |
| 18:31.34 | Samot | That's it. |
| 18:31.38 | Samot | Right there |
| 18:31.59 | shanth | ok it works now as long as nothing follows WaitExten |
| 18:32.10 | Samot | Did you press 1? |
| 18:32.18 | shanth | no |
| 18:32.20 | Samot | Show what you did |
| 18:32.28 | shanth | sure, sec |
| 18:33.10 | [TK]D-Fender | Looks like it's working fine |
| 18:33.13 | Samot | Well wait.. |
| 18:33.19 | Samot | Did you make two tests or just one? |
| 18:33.25 | shanth | http://dpaste.com/0FBK5YC |
| 18:33.34 | Samot | "as long as nothing follows" implies two tests. |
| 18:33.37 | shanth | if anything follows waitexten and it reaches timeout it just goes to the next step |
| 18:33.37 | Samot | So I want to clarify |
| 18:33.59 | [TK]D-Fender | Show us the call to match |
| 18:34.07 | shanth | works now :) i will paste call |
| 18:34.09 | Samot | ^^ |
| 18:34.16 | Samot | This is also incomplete. |
| 18:34.34 | Samot | You have nothing handling invalid DTMF responses |
| 18:34.50 | Samot | Nor do you have anything to handle valid DTMF responses. |
| 18:34.59 | Samot | So right now every response is invalid and has not place to go |
| 18:35.01 | shanth | http://dpaste.com/2RHZ7MW |
| 18:35.13 | shanth | this is just a test but yes i will add for invalid Samot |
| 18:35.40 | Samot | OK that test doesn't show a response |
| 18:35.48 | [TK]D-Fender | -- Timeout on SIP/101-00000029, going to 't' |
| 18:35.48 | Samot | So it hit the timeout |
| 18:35.51 | [TK]D-Fender | just like it should |
| 18:35.54 | shanth | my goal was to make it jump to the timeout if nothing was pressed, and it did now |
| 18:36.13 | shanth | the hindrance was having a step after WaitExten, if you put something there it ignores timeout apparently and goes to the next step |
| 18:36.16 | [TK]D-Fender | So basically yuo just finished following the instructions |
| 18:36.17 | Samot | But now let's make sure the goal of pressing something works |
| 18:36.30 | shanth | yeah let me add some stuff and try again |
| 18:36.32 | [TK]D-Fender | it doesn't |
| 18:36.38 | shanth | what? |
| 18:36.38 | [TK]D-Fender | It jumps to "t" if there IS a (t) |
| 18:36.56 | Samot | https://www.irccloud.com/pastebin/7MXgcUGp/ |
| 18:36.59 | shanth | if i put a step after WaitExten [TK]D-Fender it does not jump to t |
| 18:37.02 | Samot | That's his original dialplan. |
| 18:37.14 | [TK]D-Fender | It doesn't jump BECAUSE there is no "next priority" |
| 18:37.27 | Samot | It was hitting the Hangup() before hitting t |
| 18:37.37 | Samot | http://dpaste.com/2H891CB |
| 18:37.38 | [TK]D-Fender | IT shouldn't touch it period if there's a valid target |
| 18:37.45 | Samot | ^^ There's the call that showed it |
| 18:37.56 | [TK]D-Fender | and "t" was actually there? |
| 18:38.01 | shanth | yes |
| 18:38.05 | [TK]D-Fender | it existed, properly coded? |
| 18:38.09 | Samot | I just gave you the dialplan and the log |
| 18:38.15 | Samot | Yes it was there |
| 18:38.17 | shanth | yes because it worked if i hit a partial entry it jumped to t |
| 18:38.18 | [TK]D-Fender | so many mismatched snippits |
| 18:38.22 | [TK]D-Fender | hard to glue this together |
| 18:38.23 | shanth | lol yep |
| 18:38.32 | Samot | Look at my last two links |
| 18:38.36 | Samot | Original dialplan |
| 18:38.37 | [TK]D-Fender | Peace-meal is a crappy way to do things. |
| 18:38.38 | Samot | Original call |
| 18:38.47 | Samot | That's where it started |
| 18:38.52 | [TK]D-Fender | Anyway I suppose it feels natural to be able to continue if that's the case |
| 18:38.59 | [TK]D-Fender | so take your pick |
| 18:39.01 | shanth | yeah makes sense |
| 18:39.10 | shanth | kinda of a nice thing to know but infuriating if you dont know |
| 18:39.11 | [TK]D-Fender | also Playback() is often not a great idea for this |
| 18:39.22 | shanth | was just a test but i agree |
| 18:39.32 | Samot | Only if you want them to hear the whole thing and not interrupt it |
| 18:39.43 | [TK]D-Fender | well if there's a "t" exten what would "more lines" give you? What would be the point? |
| 18:39.48 | [TK]D-Fender | it's SUPOPSED to jump |
| 18:39.48 | shanth | Samot can you post your thing that handles 1,800 calls again? |
| 18:40.01 | [TK]D-Fender | either to match your dial, fail to do so, or timeout period |
| 18:40.49 | [TK]D-Fender | that code has nothing to do with voluem |
| 18:40.50 | Samot | https://www.irccloud.com/pastebin/nhdUPwqg/ |
| 18:40.56 | [TK]D-Fender | and I see it right there in the scroll-back |
| 18:40.56 | Samot | Just build from that |
| 18:41.20 | [TK]D-Fender | And add an invalid handler |
| 18:41.20 | Samot | That right there will timeout after 5 seconds with no response |
| 18:41.25 | Samot | Or accept 1 as a valid response |
| 18:41.30 | Samot | Work from there |
| 18:41.35 | Samot | Add the invalid handler |
| 18:41.43 | Samot | And any other "valid" matches |
| 18:44.48 | Samot | https://www.irccloud.com/pastebin/aBdZC6yJ/ |
| 18:45.09 | bananapie | Is there a dialplan app that would stop the dialplan until asterisk is done 'optimising out' local channels? WaitForAllOptimisations ? |
| 18:45.12 | Samot | For the record when there is more dialplan after WaitExten(2) (in this case) it will continue |
| 18:46.31 | shanth | thanks for the help Samot, going to lunch |
| 18:47.06 | [TK]D-Fender | bananapie, No, and that makes no sense |
| 18:56.14 | bananapie | I did this : ExecIf($["${TOLOWER(${CUT(CHANNEL,/,1)})}" = "local"]?Wait(1)) and copy/pasted it 3 times. It worked. ( I am working with origination here ) |
| 19:24.27 | [TK]D-Fender | I don't optimize out at all |
| 19:24.38 | [TK]D-Fender | I like controlling the full flow beginning to end |
| 19:27.14 | robmal | ^ serious plumber is serious |
| 19:28.14 | [TK]D-Fender | robmal, I work in a plumbing distribution company :) |
| 19:28.28 | robmal | I know ;-) |
| 19:29.03 | [TK]D-Fender | #mario |
| 19:45.03 | klow | joined 'simple_bridge' basic-bridge <- this means someone's RTP has joined a call right? |
| 19:45.20 | klow | Im trying to figure out which peers were involved in audio on a call that happened a few days ago |
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| 22:24.54 | shanth | im trying to read a balance due using SayDigits but it reads 0.00 as zero zero zero, is there an easy way to make it read that as Zero Dollars and Zero cents? |
| 22:25.38 | Samot | Use those recordings |
| 22:25.49 | Samot | There should be default recordings. |
| 22:26.02 | shanth | do i need to chop up the variable Samot? |
| 22:26.20 | Samot | You would have to look at them all and see what is there |
| 22:26.32 | Samot | and if you can use what is there instead of SayDigits |
| 22:26.38 | shanth | there is no built in good way to handle reading currency? |
| 22:26.55 | shanth | like SayCurrency? |
| 22:27.01 | Samot | No. |
| 22:27.06 | Samot | Which currency would it say? |
| 22:27.10 | Samot | In what format? |
| 22:27.16 | Samot | That's why people make language packs. |
| 22:28.50 | shanth | gotcha |
| 22:31.28 | lorsungcu | shanth: that sort of thing is easy to script outside of asterisk |
| 22:31.46 | lorsungcu | and pass back in whatever you want to read formatted how you need it |
| 22:37.27 | RovingWriter | I'd google this, but I'm not quite sure what it is called... but... some people use PTSN, some use VOiP, some use T1's or whatever.. what are those called? |
| 22:37.58 | RovingWriter | I might not be asking that question very well |
| 22:39.44 | RovingWriter | or is it purely PTSN vs VoIP? |
| 22:42.40 | wabbits | its purely pstn vs voip |
| 22:42.51 | *** join/#asterisk u0m3 (~u0m3@5-12-122-119.residential.rdsnet.ro) |
| 22:42.56 | wabbits | maybe your question is tdm vs sip? |
| 22:43.04 | wabbits | RovingWriter ^ |
| 22:43.28 | wabbits | or tdm vs rtp? |
| 22:45.20 | RovingWriter | basically, ptsn or voip are the only real options, right? |
| 22:45.31 | *** join/#asterisk u0m3 (~u0m3@5-12-122-119.residential.rdsnet.ro) |
| 22:45.40 | wabbits | even the pstn is using voip in the core now |
| 22:45.42 | RovingWriter | and things like t1, t3, stuff like that, are just the transport? |
| 22:46.31 | wabbits | pstn = public switched telephone network, regardless of technology used. |
| 22:47.28 | RovingWriter | and that is what, a big network that AT&T, southwest bell, verizon, those guys, all manage together? |
| 22:47.51 | wabbits | its all the phone companys in the world |
| 22:48.19 | RovingWriter | except the VOIP guys. but they still hand off to the PTSN people, at some point |
| 22:48.27 | wabbits | most do |
| 22:48.40 | RovingWriter | and PTSN also has to have a way to get into VOIP networks. |
| 22:48.47 | wabbits | no |
| 22:49.25 | RovingWriter | well, someone owns all the phone # blocks, and when u call a phone #, surely there is a lookup somewhere, that tells the switch where to route the call to.... yes? |
| 22:49.34 | RovingWriter | each telco has their own blocks of #'s i assume |
| 22:49.46 | wabbits | I give up |
| 22:49.50 | RovingWriter | heh |
| 22:49.52 | RovingWriter | sorry. |
| 22:50.39 | wabbits | voip = voice over Internet procotol |
| 22:50.45 | *** join/#asterisk phix (~threat@C-61-69-173-5.syd.connect.net.au) |
| 22:50.56 | wabbits | that is used inside the pstn and outside of it. |
| 22:51.21 | wabbits | the pstn is many technologies |
| 22:51.55 | wabbits | voip could be h.323 or sip or iax |
| 22:51.55 | Samot | RovingWriter: What are you trying to figure out? |
| 22:52.06 | RovingWriter | Samot, just how it works, really |
| 22:52.35 | RovingWriter | i do understand the VOIP part... |
| 22:52.52 | Samot | What work? |
| 22:52.57 | wabbits | I don't think you understand the pstn part either |
| 22:52.58 | Samot | How what works? |
| 22:53.06 | Samot | The PSTN? |
| 22:53.07 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:53.08 | RovingWriter | no, I don't understand the PSTN part either |
| 22:53.10 | Samot | VoIP? |
| 22:53.23 | Samot | Like wabbits said Public Switched Telephone Network. |
| 22:53.29 | wabbits | grabs popcorn |
| 22:53.31 | Samot | So carriers can switch calls between them. |
| 22:53.39 | RovingWriter | ok, it used to be the case that when someone wanted to make a phone call, they call an operator, and say "i want to talk to Jimmy Job in Tulsa, OK" |
| 22:53.50 | Samot | In the old days, nice ladies would answer your call and asked where you wanted to be connected to |
| 22:53.57 | RovingWriter | and the operator would physically take the copper wires out of a switchboard, and connect to Jimmy Job's |
| 22:53.58 | Samot | They where "switching" the call. |
| 22:54.10 | Samot | Now it's all done with software. |
| 22:54.18 | RovingWriter | yupyup, but now, this is done mechanically or software wise |
| 22:54.20 | RovingWriter | yea |
| 22:54.34 | Samot | What do you mean "mechanically"? |
| 22:54.41 | Samot | Wires or something is still involved. |
| 22:54.48 | Samot | Hardware is still involved. |
| 22:54.55 | Samot | Things need to be wired, connected... |
| 22:55.14 | RovingWriter | yes |
| 22:55.39 | RovingWriter | I think I get it. It's way less complicated than I had imagined. |
| 22:55.48 | Samot | Canada and the US use the North American Number Plan |
| 22:55.56 | Samot | Which denotes the 10 digit numbers |
| 22:56.15 | Samot | The NANP is run by Neustar. |
| 22:56.22 | Samot | Who control who gets what blocks |
| 22:56.24 | *** join/#asterisk mmlj4 (~mmlj4@47-48-196-90.static.gwnt.ga.charter.com) |
| 22:56.34 | Samot | Their SPID assignments |
| 22:56.39 | Samot | And the LRNs. |
| 22:57.25 | Samot | When you make a phone call the carrier looks at their internal routing.. |
| 22:57.28 | RovingWriter | ok, that's one of the big questions i had, was who controls the number blocks |
| 22:57.44 | Samot | If the number is not owned by them they lookup the LRN to find out where it routes to |
| 22:59.31 | RovingWriter | so much like how the internet works too |
| 22:59.41 | RovingWriter | not exactly, but somewhat like |
| 23:00.00 | Samot | Yes. |
| 23:00.08 | Samot | Carriers need to interconnect with each other. |
| 23:00.28 | Samot | If you're on ATT and want to call Windstream but ATT doesn't have a direct connect with them.. |
| 23:00.42 | RovingWriter | it may have to route through someone else |
| 23:00.44 | Samot | Then it has to go out some place ATT does that can send it to Windstream |
| 23:00.53 | RovingWriter | yeah, makes sense |
| 23:01.09 | Samot | It's why certain locations are cheaper than others. |
| 23:01.21 | RovingWriter | so the telco's all bill each other, and then they pass those costs on to the customer |
| 23:01.47 | Samot | Right |
| 23:01.56 | Samot | It cost money to send calls back and forth |
| 23:02.03 | Samot | Just like data on the Internet |
| 23:02.04 | RovingWriter | and voip is cheaper, because the phone companies at that point, don't have to maintain a network, really |
| 23:02.10 | RovingWriter | at least, not a big part of the network |
| 23:02.22 | Samot | It's cheaper because of the transport method. |
| 23:02.32 | Samot | The phone company no longer has to run phone wires to your house |
| 23:02.47 | Samot | They don't have to connect those wires at junction boxes... |
| 23:02.50 | Samot | Or anything else |
| 23:03.00 | RovingWriter | yeah |
| 23:03.02 | Samot | And for the most part their connection to the PSTN is over IP |
| 23:03.05 | Samot | Which is cheaper. |
| 23:03.20 | RovingWriter | and I'm not sure if this helps much, but the bandwidth is much less... 64k vs ~10k |
| 23:03.25 | RovingWriter | not sure if that plays much of a role |
| 23:03.33 | Samot | What bandwidth? |
| 23:03.38 | RovingWriter | call bandwidth |
| 23:03.46 | Samot | Depends on the codec. |
| 23:03.53 | RovingWriter | normal pots lines, from what I know, take 64k, regardless |
| 23:03.57 | Samot | ulaw is 64K before overhead |
| 23:04.11 | Samot | ulaw = standard codec for PSTN |
| 23:04.13 | RovingWriter | gotcha |
| 23:04.21 | Samot | or alaw |
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| 23:05.00 | *** part/#asterisk oalvarez (~alvarezp@pdpc/supporter/active/alvarezp) |
| 23:05.03 | RovingWriter | gotcha. i get it all better now. thanks Samot |
| 23:10.17 | RovingWriter | man Neustar also does TLD's |
| 23:10.38 | RovingWriter | seems like that company can single handedly destroy a lot of stuff haha |
| 23:11.06 | Samot | Yeah, kinda why the government is involved. |
| 23:11.21 | Samot | They answer to the FCC in the US and their counterpart in CA |
| 23:11.27 | *** join/#asterisk _carmex (~wsmith@173-31-142-124.client.mchsi.com) |
| 23:11.34 | _carmex | anyone wanna help a dummy out? |
| 23:11.48 | Samot | I thought I had been. |
| 23:11.57 | _carmex | one more dummy then ;) |
| 23:12.00 | Samot | gets RovingWriter some burn cream. |
| 23:12.29 | _carmex | i migrated from one provider to Twilio annnnnd.... i keep getting error 404. I've been googling for 2 days.... nothing |
| 23:12.37 | RovingWriter | haha ;) |
| 23:12.41 | Samot | http://weknowmemes.com/wp-content/uploads/2012/03/kelso-burn.jpg |
| 23:12.49 | RovingWriter | mean |
| 23:13.11 | Samot | _carmex: For inbound calls? |
| 23:13.41 | _carmex | outbound |
| 23:14.11 | Samot | So you try to make a call and they return a 404? |
| 23:14.17 | _carmex | yeah |
| 23:14.21 | Samot | Show it |
| 23:14.25 | Samot | asterisk -rvvvvvvvvvv |
| 23:14.28 | Samot | sip set debug on |
| 23:14.28 | _carmex | pastebin? |
| 23:14.30 | Samot | ~pb |
| 23:14.30 | infobot | somebody said pastebin was 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. |
| 23:14.39 | _carmex | ok |
| 23:14.42 | _carmex | one minute |
| 23:15.01 | _carmex | i actually have it configured for flowroute (as a test) and it still didn't work |
| 23:19.36 | _carmex | Samot, https://pastebin.com/rqFyzkPu |
| 23:20.01 | [TK]D-Fender | Looking for 2029950542 in default (domain asterisk) |
| 23:20.05 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 23:20.10 | [TK]D-Fender | Means exactly what it says |
| 23:20.21 | _carmex | yeah, but idk why it says it |
| 23:20.24 | Samot | No matching peer for '202' from 'localip:39982' |
| 23:20.33 | [TK]D-Fender | Looking for 2029950542 in default (domain asterisk) <------- |
| 23:20.51 | Samot | Contact: "202" <sip:202@localip:39982;transport=udp> |
| 23:20.52 | [TK]D-Fender | call is landing in [default] ... an you have NO MATCH |
| 23:21.02 | Samot | c=IN IP4 localip |
| 23:21.23 | [TK]D-Fender | No matching peer for '202' from 'localip:39982' <- And you don't have a peer that even matches the sender as being a declared section of your sip.conf |
| 23:21.37 | Samot | This call isn't even making past Asterisk. |
| 23:21.41 | *** part/#asterisk kharwell (kharwell@nat/digium/x-zgwhicgzdcrhgfmt) |
| 23:21.46 | Samot | It's not even attempting Flowroute or Twilio |
| 23:21.58 | Samot | It can't even auth the device making the call |
| 23:22.11 | [TK]D-Fender | well ... it IS being accepted, it jsut fails plan "B" for also having no extension defined to match it that way |
| 23:22.30 | Samot | Did you change your local IP to "localip"? |
| 23:22.33 | _carmex | okay, part of the problem may be that i scrapped the original extensions.conf |
| 23:22.40 | Samot | Because that's an issue if you didn't. |
| 23:22.42 | _carmex | Samot, yeah i took out my IPs |
| 23:22.49 | Samot | _carmex: Please don't |
| 23:22.54 | Samot | It is VERY misleading |
| 23:24.37 | _carmex | sorry. LocalIP is the softphone; asterisk is the server |
| 23:24.59 | Samot | Well |
| 23:25.10 | Samot | 7:20:51 PM S<Samot> Contact: "202" <sip:202@localip:39982;transport=udp> <-- That is misleading |
| 23:25.22 | Samot | It makes it look like the softphone isn't sending the proper information |
| 23:26.13 | _carmex | my fault. |
| 23:26.23 | _carmex | the phones and server are not on the same network |
| 23:26.48 | shanth | is there a way to make Read accept a minimum number of digits. trying to make a payment ivr and wanting to make things like expiration date a minimum of 4 digits |
| 23:28.12 | _carmex | Samot, the context part of sip.conf is where it points to [default] right? |
| 23:29.11 | RovingWriter | sooo, LRN is like DNS for phones? |
| 23:30.18 | RovingWriter | you can move the # to any datacenter, but the LRN will still tell you how to get there? |
| 23:30.28 | Samot | No. |
| 23:30.41 | Samot | You can move numbers to carriers. |
| 23:30.53 | Samot | And the LRN will tell you which carrier to send the call to. |
| 23:30.57 | RovingWriter | roger that |
| 23:32.22 | [TK]D-Fender | shanth, No. Get your input and validate it |
| 23:32.39 | [TK]D-Fender | shanth, "core show application read" <- will tell you what you can do with it. |
| 23:40.01 | _carmex | where it says "looking for xxxxx in default" what is telling it to look in default? |
| 23:40.20 | _carmex | and is default the [default] in extensions.conf? |
| 23:40.42 | [TK]D-Fender | yes |
| 23:41.06 | [TK]D-Fender | it looks there because that where that part of sip.conf told it to go |
| 23:41.17 | [TK]D-Fender | You didn't match a defined peer so it took the general setting |
| 23:41.19 | _carmex | that's what i thought, but i'm not really an expert |
| 23:41.20 | *** join/#asterisk TandyUK2 (~admin@2a02:13a0:a006:1:b4f8:8786:220b:5d0f) |
| 23:41.24 | [TK]D-Fender | ~book |
| 23:41.24 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 23:41.25 | [TK]D-Fender | ^^^^ |
| 23:41.35 | _carmex | but my sip.conf doesn't specify a [default] |
| 23:42.12 | [TK]D-Fender | no |
| 23:42.20 | [TK]D-Fender | I did not say there was a SECTION named that |
| 23:42.31 | [TK]D-Fender | that is the CONTEXT it is set to go to |
| 23:42.40 | [TK]D-Fender | context= <----------------------- |
| 23:43.25 | _carmex | right, that's what i mean |
| 23:43.34 | _carmex | context=from-phones |
| 23:43.40 | [TK]D-Fender | no |
| 23:43.45 | [TK]D-Fender | that line is clealy not being used |
| 23:43.57 | _carmex | i see that :D |
| 23:44.17 | _carmex | the only other context line is for the inbound |
| 23:44.28 | [TK]D-Fender | it's not matching ANY section you defined |
| 23:44.41 | [TK]D-Fender | Perhaps it hasn't sunk in yet... |
| 23:44.52 | [TK]D-Fender | [TK]D-Fender> You didn't match a defined peer so it took the general setting |
| 23:45.25 | [TK]D-Fender | "but I have a [fred]" <- doesn't matter, it didn't match. Anything in there means nothing to this call. |
| 23:45.53 | [TK]D-Fender | it falls to [general] as it's even being accepted without a match (which is pretty insecure) |
| 23:48.10 | *** join/#asterisk TandyUK (~admin@2a02:13a0:a006:1:b4f8:8786:220b:5d0f) |
| 23:51.28 | _carmex | idk what just happened, but it looked like it was flooded and now i'm getting 403 Forbidden :| |
| 23:53.58 | [TK]D-Fender | Flood wouldn't have anything to do with that |
| 23:54.02 | [TK]D-Fender | not match is no match |
| 23:54.40 | _carmex | yeah, sorry. i'm not really tracking. i'm not exactly sure what you mean by "it" when you say it's not matching |
| 23:57.26 | [TK]D-Fender | * looks at the caller and goes "I don't KNOW you" |
| 23:59.30 | wabbits | _carmex the inbound call does not match anything in the dialplan. |
