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| 01:41.31 | TSM | has anyone seen problems with SIP registrations where provider is using IPv6 and your device is IPv4? |
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| 01:43.51 | drmessano | Why would that fail? |
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| 01:50.52 | TSM | i have no idea, I am trying to install a poly phone at someones house, it was working fine as it is just their office unit, when when in their house it just fails all the time, I have taken it back to the office and it now works fine, http://pastebin.com/MPcyq7SB , i checked the permit setting which is on 0.0.0.0 which is odd, i tried disabling SIP ALG on the router and no problem, they are |
| 01:50.52 | TSM | on a TWC NY connection |
| 01:53.48 | TSM | it works fine with xlite which is weird though |
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| 02:12.16 | drmessano | That doesnt even remotely say "IPv6" |
| 02:14.26 | drmessano | Every one of your registers is faling |
| 02:14.34 | drmessano | Check the credentials |
| 02:14.46 | drmessano | "Ive taken it back to the office" |
| 02:15.00 | drmessano | I dont know what variable that changes |
| 02:15.13 | drmessano | Where is Asterisk located? What does moving it change? |
| 02:22.27 | TSM2 | the server is hosted with rentpbx, the office has a pure IPv4 connection while the guys house is IPv6/IPv4, i litrally picked up the phone and took it back to the office and connected it up and it now connects to the * server |
| 02:26.53 | TSM2 | could there be some carrier grade natting going on and just broken |
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| 14:19.41 | *** join/#asterisk infobot (ibot@69-58-76-73.ut.vivintwireless.net) |
| 14:19.41 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.5.0 (2015/08/07), 11.19.0 (2015/08/07), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 14:31.10 | newtonr | Jesterboxboy, there might be something in console or logs if you turn up verbose and debug |
| 14:31.34 | Jesterboxboy | both to 10 but nothing |
| 14:31.43 | newtonr | Jesterboxboy, check out logger.conf and verify you have all various log channel types going where you want |
| 14:32.01 | Jesterboxboy | but i should see it in the console ? |
| 14:32.04 | Jesterboxboy | if there is something |
| 14:32.22 | newtonr | Jesterboxboy, if you have those log channels routed to the console... |
| 14:33.40 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration |
| 14:33.55 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Logging |
| 14:36.57 | Jesterboxboy | thanks solved it |
| 14:37.31 | newtonr | solved the logging issue or the cdr issue? |
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| 15:16.44 | Get_The_Fish | Hey can someone help me get inbound calls working on this config? I've put a couple full days into this config and this is the last thing that I can get figured out. http://pastebin.com/ADeGSNde |
| 15:22.04 | file | Get_The_Fish, Flowroute is not including line information in the incoming INVITE, as a result you will need to do IP based matching |
| 15:25.11 | Get_The_Fish | file: even that isnt working. And IP matching is far less than ideal, and not something we had to do in chan_sip. |
| 15:27.04 | file | IP based matching does work, if the right IP address or range is used |
| 15:27.10 | file | and did you use anonymous calling in chan_sip? |
| 15:30.30 | Get_The_Fish | http://pastebin.com/FZEh14fz |
| 15:30.48 | Get_The_Fish | file: no, I didnt. And here is a paste showing it not working. |
| 15:32.08 | file | the auth= option in an endpoint specifies inbound authentication |
| 15:32.31 | file | for outbound authentication use outbound_auth= |
| 15:35.20 | Get_The_Fish | file: Removed auth from the endpoint record. Outbound_auth is in the registration section. |
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| 15:36.14 | [TK]D-Fender | Should be in the endpoint |
| 15:39.26 | Get_The_Fish | file: here is a paste with the same provider's chan_sip configuration and a packet trace of an inbound call. |
| 15:39.34 | Get_The_Fish | file: http://pastebin.com/x8xVzXha |
| 15:39.47 | file | that's doing IP based matching |
| 15:40.51 | Get_The_Fish | file: I dont see an IP address in that configuration file, do you?? |
| 15:41.00 | [TK]D-Fender | <PROTECTED> |
| 15:41.01 | [TK]D-Fender | ^^^ |
| 15:41.05 | file | I see a hostname which resolves to an IP address |
| 15:41.10 | [TK]D-Fender | I resolves that you know.... |
| 15:41.14 | [TK]D-Fender | it* |
| 15:41.19 | Get_The_Fish | [TK]D-Fender: Not according to pjsip.conf |
| 15:41.40 | [TK]D-Fender | contact supports fqdn.... |
| 15:42.01 | file | you pasted a sip.conf configuration which is used by chan_sip, that resolves the hostname upon load |
| 15:42.07 | [TK]D-Fender | matching the host.... as opposed to username, "to", etc |
| 15:42.08 | Get_The_Fish | file: big difference between a HOSTNAME and an IP. Big. match requires IP, which I have in the configuration, by the way. Still. Not. Working. |
| 15:42.26 | [TK]D-Fender | Get_The_Fish: Show your new config.... |
| 15:42.38 | file | match doesn't require an IP address, it will resolve a hostname to all A records |
| 15:42.45 | file | unlike chan_sip which resolves to only 1 |
| 15:42.52 | Get_The_Fish | [TK]D-Fender: The difference is that sip.flowroute.com could be several IPs', as it is a SRV record. |
| 15:43.14 | Get_The_Fish | file: the docs specifically state that match requires an IP. |
| 15:43.20 | [TK]D-Fender | While file is now saying it will resolve all of them |
| 15:43.37 | [TK]D-Fender | an IP that matches.... and it will resolve all the "A"'s as he says |
| 15:43.48 | [TK]D-Fender | Let's see you current config |
| 15:44.13 | file | the hostname support got added later |
| 15:44.19 | file | create issue to update doc |
| 15:44.47 | mjordan | Documentation: the task that never ends. |
| 15:45.39 | Get_The_Fish | [TK]D-Fender: http://pastebin.com/FZEh14fz |
| 15:45.41 | newtonr | mjordan, someone started writing it not knowing what it was... this is the documentation that never eeendds! *muppet lambs dance around* |
| 15:45.54 | [TK]D-Fender | [flowroute] |
| 15:45.55 | [TK]D-Fender | type=auth |
| 15:45.58 | [TK]D-Fender | you still left this in |
| 15:46.07 | [TK]D-Fender | he just told you to get rid of it for the "outbound_auth" options |
| 15:46.19 | file | not quite |
| 15:46.38 | file | the "auth=flowroute" needs to be removed from the flowroute endpoint |
| 15:46.48 | file | it is challenging Flowroute for authentication which they won't provide |
| 15:46.49 | [TK]D-Fender | that's what' I was saying |
| 15:46.58 | [TK]D-Fender | perhaps not the most explicit workding.. |
| 15:47.06 | [TK]D-Fender | I did mean removal of all reference to it |
| 15:47.11 | [TK]D-Fender | wording* |
| 15:47.16 | [TK]D-Fender | can't type today... |
| 15:47.29 | file | it's needed for outbound authentication |
| 15:49.35 | [TK]D-Fender | ah, just remove from the endpoint |
| 15:50.02 | file | yup yup |
| 15:57.23 | qakhan | i am using Page() app to call from p1 to 2 phones p2 and p3. when both phones p2 and p3 hangup the call, p1 does not hangup the call. |
| 15:58.16 | qakhan | call still going on on p1 |
| 16:01.53 | Get_The_Fish | file: Hey thanks for your help, that is working now with hostname instead of IP. Now I just get to deal with some NAT. |
| 16:02.14 | rmudgett | qakhan: p1 controls the Page conference. |
| 16:02.23 | rmudgett | It must hangup. |
| 16:02.44 | Get_The_Fish | Do we have a list of channel variables with PJSIP? |
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| 16:07.00 | qakhan | rmudgett here is scenario what i am trying to accomplish. there are 2 people working in 2 different stores. backoffice guys need to communicate with them using 1 ext (i am using 8000). |
| 16:07.56 | qakhan | when they dial 8000 both phones p2 and p3 auto answer the call and back office guys call person name who they want to talk. |
| 16:08.41 | qakhan | desired person answer (i am here) and other person hangup the call. |
| 16:11.52 | qakhan | or is there any app in asterisk 11.10 which take person name (caller says person name) and asteriks will dial that person extension |
| 16:13.56 | rmudgett | Asterisk has no built in voice recognition |
| 16:14.19 | [TK]D-Fender | 3rd party via sphinx, lumenvox, etc |
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| 16:18.36 | SpeakerToMeat | Hello. it's me again :D |
| 16:20.13 | SpeakerToMeat | Quick question, so, you want to get a sub $80 sip phone on amazon (or a good retailer for voip), that at least has: a) tls/srtp b) a good sized directory c) good asterisk integration d) a not too confounding UI |
| 16:20.37 | SpeakerToMeat | What would you go for. Grandstream (simpler model, no color screen), cisco (simpler model), or another brand? |
| 16:21.01 | SpeakerToMeat | 1 sip line only required (2 lines would be a plus, is not a must) |
| 16:21.12 | [TK]D-Fender | ~gs |
| 16:21.12 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 16:21.15 | [TK]D-Fender | ~grandstream |
| 16:21.15 | infobot | somebody said grandstream was the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 16:21.28 | SpeakerToMeat | Ok |
| 16:21.41 | SpeakerToMeat | I get it. |
| 16:21.43 | SpeakerToMeat | :) |
| 16:22.08 | SpeakerToMeat | I see therealcircuit was ousted. Or eaten |
| 16:23.21 | qakhan | [TK]D-Fender can i call to 2 phones same time and both phones auto answer same time. without using page () |
| 16:24.01 | SpeakerToMeat | What about cheap line cisco? My old and only experience with cisco voips was that trying to make it do anything with a non CCD setup required blood donations |
| 16:24.10 | igcewieling | qakhan: unlikely |
| 16:24.43 | SpeakerToMeat | ccd? cca? meh I've forgoten most of that hell already |
| 16:25.14 | [TK]D-Fender | qakhan: Dial dials. Page dials. That's just about it. |
| 16:25.29 | SpeakerToMeat | Btw for my own use, what's the cheapest 1 fxo 1 fxs gateway out there that is usseable? |
| 16:25.34 | [TK]D-Fender | qakhan: How you go about doing that is up to you |
| 16:25.59 | [TK]D-Fender | SpeakerToMeat: Linksys SPA-3102 or whatever the new SPA version # is |
| 16:26.16 | SpeakerToMeat | [TK]D-Fender: Thanks |
| 16:26.18 | igcewieling | SpeakerToMeat: three rules of life: variables dont. constants aren't. cheap voip stuff is crap. |
| 16:26.52 | qakhan | [TK]D-Fender Dial() dials both phones but only 1 phone auto answer the call. |
| 16:27.04 | qakhan | while Page() does both |
| 16:27.11 | [TK]D-Fender | The SPA is very inexpensive and generally "good enough" for light us |
| 16:27.30 | [TK]D-Fender | qakhan: Correct. |
| 16:27.38 | igcewieling | qakhan: why don't you want to use page? |
| 16:28.02 | SpeakerToMeat | [TK]D-Fender: Any preference regarding the sip phone? Other than run away from grandstream? |
| 16:28.52 | igcewieling | SpeakerToMeat: any preferences we have would be outside your "cheap" requirement. For example, I only use Polycom phones. |
| 16:28.59 | qakhan | i want call to be hangup if called party (p2 or p3) hangup the call. |
| 16:29.06 | SpeakerToMeat | igcewieling: I see. |
| 16:29.39 | SpeakerToMeat | igcewieling: besides the price range, also I have something with polycom. It's stupid, I know... but... I feel like I'd erupt in flames if I had to buy one |
| 16:29.57 | igcewieling | qakhan: perhaps someone else can think of a way, but the only way I can think of to accimplish what you want to accomplish is a complicated setup of scripts and dialplan. |
| 16:30.43 | [TK]D-Fender | SpeakerToMeat: You wanted our opinions. If you have irrational fears like that then we wish you the best of luck. |
| 16:30.59 | SpeakerToMeat | [TK]D-Fender: It's ok, I take the opinions in anyhow. thanks |
| 16:35.40 | SpeakerToMeat | Is polycom's quality homogeneous thorough their line? including cheaper phones? I found a soundpoint ip 330 for a good price |
| 16:36.07 | [TK]D-Fender | I'd recommend a model is wasn't discontinued years ago |
| 16:40.50 | Get_The_Fish | SpeakerToMeat: I've been happy with my digium phones so far. |
| 16:45.31 | qakhan | did anyone try this google Speech recognition |
| 16:45.53 | [TK]D-Fender | qakhan: Yes, I'm sure thousands of people have googled that |
| 16:47.25 | qakhan | i meant to ask is it successful ? |
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| 16:48.31 | [TK]D-Fender | qakhan: Google searches tend to be as successful as the user is smart at actually providing useful terms to search by. |
| 16:49.10 | Get_The_Fish | So do we have a pjsip equivalent of ${SIPPEER}? |
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| 17:00.10 | *** join/#asterisk airjump (~Thunderbi@p20030070CE1541D6D1DF8105E395654C.dip0.t-ipconnect.de) |
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| 17:02.28 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:02.56 | Get_The_Fish | Is there a PJSIP channel variable similar to ${SIPPEER} ?Sorry I was disconnected, didnt see an answer. |
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| 17:10.33 | litn | hey, so in my AGI script, I ask them to press some keys for different options that will transfer the call to certain people. When I transfer I use the Dial application. Is there a better way to do this? |
| 17:10.38 | litn | or is that proper for a menu? |
| 17:11.07 | litn | just want to make sure because we have had some weird issues since rolling out the menu and I was thinking it might be since the call doesn't really transfer but dials out |
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| 17:15.06 | [TK]D-Fender | Do not abuse the term "transfer" |
| 17:15.12 | [TK]D-Fender | you are processing a call in the dialplan |
| 17:15.21 | [TK]D-Fender | this is not 2 parties talking to one another |
| 17:15.33 | [TK]D-Fender | you are not transferring away from someone |
| 17:15.51 | litn | oh I see |
| 17:15.53 | litn | I understand |
| 17:16.07 | [TK]D-Fender | So just like a standard Waitexten based prompt, etc, this is just a dumb dial like any other if you are simply calling a normal peer on your system |
| 17:18.16 | litn | I'm seeing some weird stuff happen, I don't know the right terms but it seems like ghost calls, |
| 17:18.38 | litn | someone will press the option for reception, go to that queue, talk to someone and hang up, and then reception tell me that same number rings them again but noone is there |
| 17:19.29 | [TK]D-Fender | We'd actually need something real to look at for that issue |
| 17:19.30 | litn | only started happening when I put this dial plan in place |
| 17:19.57 | litn | maybe I can post the agi script and that part of my dial plan and you can tell me if I am missin anything obvious, this is my first time writing an agi script or even custom dialplan |
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| 17:27.10 | litn | [TK]D-Fender: Here it is.. http://pastie.org/10448625 |
| 17:27.33 | litn | [TK]D-Fender: I use multithreading so I can run the http call to look the customer up in our db *while* it is streaming the initial greeting btw |
| 17:27.52 | litn | the dialplan part is in triple quotes at the top |
| 17:29.04 | litn | it works well in my testing but yeah there are some weird ghost calls and stuff |
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| 17:30.59 | litn | [TK]D-Fender: and then also I wrap everything in a catch all - sloppy as that may be - just in case there is any ege case scenario, so I send the person to reception in any instance like that |
| 17:31.00 | [TK]D-Fender | 'Local/951@ext-queues' |
| 17:31.20 | [TK]D-Fender | dialing local channels causes the 2 channels to break off from where they were before and probably causes issues with your AGI |
| 17:31.35 | [TK]D-Fender | Some of the basics you should know on how they get masquearded out |
| 17:31.42 | [TK]D-Fender | masqueraded* |
| 17:31.48 | litn | since this is dialplan, it should instead be like a goto ? |
| 17:32.11 | [TK]D-Fender | calling Dial like that has its consequences |
| 17:32.20 | [TK]D-Fender | Doing a Goto will imply having to leave your AGI |
| 17:32.39 | [TK]D-Fender | Is that stream the only point of using AGI there? |
| 17:33.34 | [TK]D-Fender | Because that would be a blocking action... |
| 17:34.05 | litn | no, |
| 17:34.10 | litn | AGI blocks yes |
| 17:34.12 | litn | but it's in a separate thread |
| 17:34.18 | litn | I call http right after I start that thread |
| 17:34.26 | litn | so while it streams, I'm fetching JSON that has info about this customer |
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| 17:35.05 | litn | [TK]D-Fender: once they hit a number and go somewhere, nothing else needs to happen on that agi script |
| 17:35.08 | litn | it can exit |
| 17:35.49 | [TK]D-Fender | Then you should set the AGI exit point and just ext. |
| 17:35.52 | [TK]D-Fender | exit* |
| 17:36.26 | litn | googling exit point didn't show anything- do you mean to call Goto? |
| 17:38.53 | [TK]D-Fender | read your AGI basics where you get to set where the dialplan resumes upon exit |
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| 18:03.27 | Get_The_Fish | So now I'm getting this: No SIP authenticator registered. Assuming authentication is not required when starting Asterisk |
| 18:04.36 | Get_The_Fish | (as a console warning message) |
| 18:05.25 | WIMPy | Time to load some modules. |
| 18:06.54 | litn | [TK]D-Fender: thanks for your guidance! I changed it to set context, set extension, sert priority, and then exit rather than dial :) |
| 18:06.57 | litn | we'll see if the issues go away |
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| 18:11.39 | Get_The_Fish | WIMPy: Do I have a module that isn't loaded?? |
| 18:11.56 | WIMPy | At least one. |
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| 18:18.14 | igcewieling | Get_The_Fish: did you mess with /etc/asterisk/modules.conf? |
| 18:18.52 | Get_The_Fish | Quite a bit, actually. However I had autoload on, and I'm only noloading stuff I'm not using. |
| 18:22.52 | igcewieling | Get_The_Fish: step 1: stop noloading stuff. |
| 18:23.24 | igcewieling | if that helps you can track down which specific module you need. |
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| 19:14.31 | Get_The_Fish | Anyone have a recommendation for the best way to make IVR recordings? |
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| 19:16.15 | Get_The_Fish | IE software on a PC, phone, and IVR that makes recordings... |
| 19:25.24 | [TK]D-Fender | Grab phone. Make recording |
| 19:25.59 | [TK]D-Fender | great because you already have everything you need and it goes directly into the right format |
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| 20:01.08 | Get_The_Fish | [TK]D-Fender: Didnt know if there was an easier way. |
| 20:01.51 | [TK]D-Fender | What's easier than picking a phone you already have and recording with software you already hav installed? |
| 20:02.10 | Get_The_Fish | I could use some help with no audio now on inbound calls. It APPEARS that asterisk is doing what it's supposed to be doing, and sending audio back to the carrier, however I hear nothing. |
| 20:02.17 | igcewieling | You could run an audo capture program like Audacity and a microphone, then postprocess the file into the correct format. Using a phone sounds easier. |
| 20:02.39 | SpeakerToMeat | Is there a way I can check whether a trunk ( a chan_sip) is using tls/srtp? |
| 20:02.44 | Get_The_Fish | [TK]D-Fender: Yeah, kinda does. Making the dialplan to do it sounds like a pain |
| 20:03.37 | Get_The_Fish | http://pastebin.com/EvvD5AVk for the no audio issue |
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| 20:04.43 | [TK]D-Fender | Get_The_Fish: pain? about half a minute. |
| 20:04.52 | [TK]D-Fender | Get_The_Fish: Less time than it took to ask |
| 20:05.47 | igcewieling | Get_The_Fish: um, call yourself, leave your message a voicemail, copy it to where you need it |
| 20:06.01 | Get_The_Fish | [TK]D-Fender: Actually I just found a dialplan snippet that will do it. |
| 20:06.17 | Get_The_Fish | [TK]D-Fender: *slaps forehead* |
| 20:06.18 | [TK]D-Fender | all of 1 line |
| 20:06.27 | [TK]D-Fender | Everything else is gravy |
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| 20:12.24 | SpeakerToMeat | Will using freepbx/elastix to begin turn me into a bad user? |
| 20:15.42 | [TK]D-Fender | Depends on you |
| 20:15.55 | SpeakerToMeat | Ok. I get it. |
| 20:16.16 | SpeakerToMeat | Regarding the sip connection? any tips please? |
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| 20:16.34 | [TK]D-Fender | If you figure you'll be as lazy as humanly possible then it's a great first step. Pile on a little sense of entitlement and you're well on your way. |
| 20:16.42 | SpeakerToMeat | I have a sip trunk setup encryption=yes is there any easy way to check the connection is actually tls and srtp? |
| 20:17.12 | [TK]D-Fender | it'll get REFUSED if they don't agree |
| 20:17.21 | [TK]D-Fender | And you can tell by looking at the actual call debug |
| 20:17.32 | SpeakerToMeat | Ah since encryption=yes makes it mandatory |
| 20:19.17 | [TK]D-Fender | yes != maybe |
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| 20:26.17 | [TK]D-Fender | heads home... |
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| 21:04.02 | *** join/#asterisk jkroon (~jkroon@196.36.167.46) |
| 21:07.00 | jkroon | hi all, after some debugging on FAXOPT(faxdetect)=yes I realized that any frame that is not ulaw or alaw simply gets skipped for fax detection from the framehook handler - is there any way to in the framehook enable transcoding/decoding of the voice data into SLINEAR and rather process that? Is there documentation I can look at for writing the code? We would like to enable faxdetect, unfortunately the upstream codec is g729. once we detect a fax we |
| 21:07.00 | jkroon | can force a switch to t38 (which based on my testing should work but obviously the detection needs to work first). |
| 21:07.07 | jkroon | this is in res/res_fax.c |
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| 21:08.25 | WIMPy | I think you are better off in -dev with that question. |
| 21:09.11 | jkroon | that probably be true. how are you? |
| 21:10.01 | WIMPy | Somewhat tired, otherwise ok. |
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| 21:24.50 | Jesterboxboy | is it possible to use a variable in the pattern matching part in a dialplan? |
| 21:25.09 | Jesterboxboy | exte => _${variable},1, do something :? |
| 21:25.39 | [TK]D-Fender | no |
| 21:25.53 | Jesterboxboy | okay thank you |
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| 21:45.08 | *** join/#asterisk pjensen00 (~per@ip-69-178-218-71.far.ideaone.net) |
| 21:46.49 | pjensen00 | Greetings! I'm doing a 'redirect' in ARI. 1/2 the time I get asterisk sending our SIP server a 302 with the information I provide. The other half of the time I am seeing Asterisk sending a 603. Is there some kind of obvious failure scenario on a redirect a 603 would be triggered? |
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| 21:50.03 | WIMPy | Jesterboxboy: Yes, but the variable has to be defined in the dialplan as global. |
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| 22:07.37 | pjensen00 | @WIMPy was that at me? |
| 22:08.03 | WIMPy | no |
| 22:08.09 | pjensen00 | Oh. Ok. |
| 22:08.38 | jkroon | WIMPy, working FAXOPT(faxdetect)=yes in CNG for non-G.711 channels :D |
| 22:09.08 | WIMPy | jkroon: You changed the channel? |
| 22:09.19 | jkroon | "minor" patch to res/res_fax.c |
| 22:09.53 | WIMPy | Cool. I would think that others would be interested as well. |
| 22:09.56 | jkroon | basically drop the set read format code that was there for faxdetect that didn't work anyway and replaced it with a translator path when needed. |
| 22:10.05 | WIMPy | Even if I can't remember anyone asking for it. |
| 22:11.22 | jkroon | very useful - use case: incoming SIP, happens to be a fax call, outbound to phone ... incoming side doesn't initiate switch to t38, and obviously neither does the phone, now FAXOPT(faxdetect)=yes kicks in, drop the "peer" channel and sends to the fax extension instead. |
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| 22:14.30 | jkroon | WIMPy, agreed - just not sure where to post patches nowadays. |
| 22:14.40 | jkroon | bug tracker or reviewboard? |
| 22:14.58 | WIMPy | That would be gerrit now. |
| 22:15.08 | jkroon | oi, gerrit.asterisk.org? |
| 22:15.10 | WIMPy | Theres a page on the wiki on how to put somethign up on gerrit. |
| 22:15.27 | rmudgett | Create an issue as well |
| 22:15.37 | jkroon | rmudgett, ok. |
| 22:15.46 | WIMPy | It was rather daunting for someon who might need it once a year or less. :-( |
| 22:16.12 | jkroon | would logins from the old reviewboard pull through automatically? |
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| 22:22.02 | jkroon | WIMPy, would need to refactor the patch for master. |
| 22:22.12 | jkroon | will only apply as is to the 11 series. |
| 22:22.29 | jkroon | minor changes I reckon but I'm not going to make them tonight. |
| 22:22.37 | WIMPy | is stuck on 11 as well. |
| 22:24.20 | rmudgett | jkroon: If you have an accepted signed contribution license then you can log into gerrit with your JRIA login. |
| 22:27.06 | jkroon | rmudgett, i do, let me just refactor the patch for master/ first. I'll obviously have to submit two patches, one for 11.X and one for master/ so may just as well do both in one go. |
| 22:27.41 | rmudgett | And a cherry pick for v13 |
| 23:01.28 | *** join/#asterisk moe` (~moe@node-be028882.scarlet.cw) |
| 23:01.31 | moe` | hey guys |
| 23:01.36 | moe` | another stupid problem |
| 23:01.47 | moe` | bria client to bria client, no audio |
| 23:02.12 | moe` | there are other bria and linphone clinets where it works fine, just the one client has no audio |
| 23:02.19 | moe` | but they can call landlines and voicemail |
| 23:02.22 | moe` | and it works |
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| 23:18.14 | t4nk997 | hi, my team and i have created a real-time payments platform for asterisk - anyone interested in trying it? |
| 23:28.14 | TazzNZ | so you pay me to use asterisk. in real time ? :D |
| 23:28.35 | t4nk997 | ha :D |
| 23:28.55 | *** part/#asterisk kharwell (kharwell@nat/digium/x-bvvxtewnoiajrift) |
| 23:29.10 | t4nk997 | between carriers, the payments happen as the call happens. there is no pre-payment needed. |
| 23:34.44 | igcewieling | Between carriers payments are post pay. |
| 23:35.06 | igcewieling | generally after signing a 100 page contract and an credit check. 8-) |
| 23:37.05 | t4nk997 | ya there is def some post pay, but what a waste of time (100 pg contract?!) and credit check / legal / risks costs built into the price |
| 23:38.18 | igcewieling | Sorry, but in my view a carrier who gets their mins as prepay is not, in my view, a carrier. |
| 23:38.43 | igcewieling | I'm sure others would disagree. |
| 23:39.09 | t4nk997 | how long do you think this industry can function (business-wise) as it does? |
| 23:39.29 | igcewieling | t4nk997: as long as they can extract money from customers. |
| 23:39.49 | t4nk997 | seems to be a race to the bottom, where no one trusts each other - fair? |
| 23:40.19 | igcewieling | I don't expect Verizon or Level 3 to trust me. I don't t trust them. |
| 23:40.54 | t4nk997 | absolutely, coupled with the costs that lack of trust adds to the price and constrains margins |
| 23:41.56 | igcewieling | We'll have to agree to disagree. |
| 23:42.52 | t4nk997 | disagree on what part? thought we agreed there is lack of trust. :) |
| 23:43.46 | igcewieling | I think a lack of trust is good. You seem to think it is bad. |
| 23:45.22 | t4nk997 | no no - we agree - no one should trust anyone in this space. i have a solution where no one needs to trust anyone (incl ridding of pre or post paying) through real-time payments. |
| 23:46.06 | t4nk997 | and when neither side needs to trust the other, that risk is eliminated |
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