| 00:14.12 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 00:22.40 | [TK]D-Fender | The ACA isn't even "socialized medicine" |
| 00:24.01 | [TK]D-Fender | The insurance mandate is basically the republican RomneyCare, which is pure corporatism, not socialism |
| 00:24.33 | [TK]D-Fender | Obama was a spineless sell-out in going for it in the first place. |
| 00:25.29 | [TK]D-Fender | Now some of the protections offered around that mandate were good, but the financial tie-in was a giant failure ticking like a bomb and the providers are now "changing the deal" |
| 00:25.53 | [TK]D-Fender | https://www.youtube.com/watch?v=WpE_xMRiCLE |
| 00:26.10 | *** join/#asterisk monsterco (~monsterco@toroon474aw-lp130-02-70-50-209-133.dsl.bell.ca) |
| 00:29.00 | *** part/#asterisk monsterco (~monsterco@toroon474aw-lp130-02-70-50-209-133.dsl.bell.ca) |
| 00:38.08 | drmessano | Yep, which is why when people have freaked out and gone off about trump repealing Obamacare, my thought is "you mean at broken ass piece of legislation that protected nothing and was ripe to be replaced?" |
| 00:38.19 | drmessano | That* |
| 00:38.44 | drmessano | It was a deal with the devil that had no foundation. |
| 00:40.39 | [TK]D-Fender | More fun : http://www.independent.co.uk/news/world/americas/us-elections/donald-trump-obamacare-repeals-latest-policies-quote-replacement-president-elect-a7412621.html |
| 00:40.46 | [TK]D-Fender | "Donald Trump: I may not repeal Obamacare, President-elect says in major U-turn" |
| 00:41.20 | drmessano | You're going to base your entire reason to be angry at something that couldn't stay the rest of a changing administration? ROFLCOPTER |
| 00:41.34 | drmessano | Test |
| 00:41.56 | drmessano | He won't repeal it |
| 00:42.11 | drmessano | Fix it, gut it. Repair it. Maybe. |
| 00:42.23 | [TK]D-Fender | It actually doesn't deserve to exist. It put the corporations in front. |
| 00:43.08 | drmessano | Right. We're going to hand you everyone's asses and make it a mandate, as long as you promise to cover everybody |
| 00:43.10 | [TK]D-Fender | 2 solutions : use regulations to protect the consumer as a product and go private again with full choice. Or go PURE single-payer |
| 00:43.11 | drmessano | Some deal |
| 00:43.19 | [TK]D-Fender | The half-way is BS |
| 00:44.20 | [TK]D-Fender | #2: the rest of the industrialized PLANET agrees is the way to go |
| 00:48.26 | *** join/#asterisk monsterco (~monsterco@toroon474aw-lp130-02-70-50-209-133.dsl.bell.ca) |
| 00:49.17 | monsterco | is pjsip DNS Resolver feature a security replacement for something like VPN? |
| 00:49.21 | monsterco | or close to it? |
| 00:50.46 | drmessano | What? |
| 00:51.37 | drmessano | A DNS resolver resolves host names to IP addresses using DNS |
| 00:51.50 | drmessano | A VPN is, well, a VPN |
| 00:55.07 | [TK]D-Fender | How is DNS being compared to ... an encryption tunneling protocol? |
| 00:55.29 | [TK]D-Fender | Is my car stereo a replacement for JELLO? |
| 00:55.50 | drmessano | You can eat both |
| 00:55.54 | drmessano | One is less messy |
| 00:56.07 | drmessano | And more solid |
| 01:12.43 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 01:14.21 | *** join/#asterisk freebs (~freebs@unaffiliated/freebs) |
| 01:26.35 | Mango45 | What do you all do for simultaneous ringing to cell phones? |
| 01:27.10 | [TK]D-Fender | Try qualifying that into an answerable question |
| 01:27.28 | [TK]D-Fender | that isn't a thing to have an opinion on as worded |
| 01:27.46 | [TK]D-Fender | I do nothing. The cell phones are ringing. I like it when phones can ring. |
| 01:27.48 | [TK]D-Fender | The End |
| 01:28.18 | [TK]D-Fender | Where's the objective or issue in there? |
| 01:28.20 | Mango45 | Ok. Dial(SIP/DeskPhone&SIP/15551234567@outbound) is about the simplest way to ring multiple phones. Do any of you do this in more complicated/better ways? |
| 01:28.50 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 01:32.43 | [TK]D-Fender | Multiple devices... |
| 01:33.02 | [TK]D-Fender | before you ask for "better".... what is this NOT doing for you? |
| 01:33.45 | [TK]D-Fender | In order to give us a hint about why you're even asking this we'd assume it FAILS your goals in some way |
| 01:34.01 | Mango45 | I haven't decided what my goals are yet. Hence wanting to know what other people do. |
| 01:34.07 | [TK]D-Fender | That does it. |
| 01:34.15 | [TK]D-Fender | I want 2 things ringing.. that'll do it |
| 01:36.14 | Mango45 | Ok. Do I want Asterisk to do anything to my cell phone other than ring? Detect cell voicemail and not connect the call? Ring the cell after a delay to give me time to pick up my office phone? |
| 01:36.29 | [TK]D-Fender | Do you? |
| 01:37.16 | Mango45 | Should I be doing something else that I haven't thought of? |
| 01:37.30 | [TK]D-Fender | Do you maybe do 2 separate dials to just dial your desk FIRST .... and then stop and try something different? |
| 01:37.56 | [TK]D-Fender | Don't go taking my request for you to define your goal and then ask YOURSELF the question ... and not answer it |
| 01:38.16 | [TK]D-Fender | Find some imagination and make up your mind about what you want. |
| 01:38.27 | [TK]D-Fender | You can't solve a puzzle you can't define |
| 01:38.34 | Mango45 | So the question is, "What can Asterisk do that I haven't thought of?" |
| 01:38.39 | Mango45 | (In this context.) |
| 01:38.43 | [TK]D-Fender | We are psychic |
| 01:38.51 | [TK]D-Fender | Now pick a quesiotn that doesn't require magic |
| 01:38.55 | [TK]D-Fender | aren't* |
| 01:39.04 | [TK]D-Fender | We don't know what you haven't thought of. |
| 01:39.22 | Mango45 | That's why I'm asking what people here do. |
| 01:39.33 | [TK]D-Fender | It does what I want |
| 01:39.41 | [TK]D-Fender | I want 2 things to ring, just like you said |
| 01:40.17 | [TK]D-Fender | This is a stupid guessing game |
| 01:40.41 | Mango45 | So STFU if you don't want to answer. |
| 01:40.47 | [TK]D-Fender | Dial() <- it can dial multiple things. Who DOESN'T use dial to dial thing? |
| 01:41.12 | [TK]D-Fender | What are you going to do, shove them in a Queue? Does that make sense for your needs? |
| 01:41.20 | [TK]D-Fender | Does Page() make sense? |
| 01:41.28 | Mango45 | I'll look it up; thanks. |
| 01:41.33 | [TK]D-Fender | How dialplan apps can YOU name that will cause a call to be placed? |
| 01:41.57 | [TK]D-Fender | But you don't even have a goal |
| 01:42.13 | [TK]D-Fender | We have to first guess what you thought of ... and then guess what you'd WANT. |
| 01:42.41 | [TK]D-Fender | This isn't "Ms. Cleo" |
| 02:07.45 | Samot | Heres a vague thought... |
| 02:07.53 | Samot | Please help with it. |
| 02:09.36 | [TK]D-Fender | Wait ... you haven't confirmed that I have to know everything in your head first.... |
| 02:09.48 | [TK]D-Fender | And then guess your needs or wants |
| 02:10.01 | [TK]D-Fender | Sorry, could you be a little less specific? |
| 02:14.27 | Samot | You know. Stuff. Stuff about things. The SoT |
| 02:31.30 | *** join/#asterisk freebs (~freebs@unaffiliated/freebs) |
| 02:33.50 | [TK]D-Fender | And you forgot to answer that in the form of a question. |
| 02:33.53 | [TK]D-Fender | #trebek |
| 02:46.08 | *** join/#asterisk rpifan (~rpi@73.106.75.17) |
| 02:58.41 | *** join/#asterisk freebs (~freebs@unaffiliated/freebs) |
| 03:13.05 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 03:46.28 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
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| 04:54.18 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 05:16.49 | drmessano | Geesh.. you were a little hard on him |
| 05:16.53 | drmessano | It's not like he voted for Cisco |
| 05:17.02 | drmessano | #couldntresist |
| 05:17.13 | Mango45 | :) |
| 05:17.13 | drmessano | #DividedDialplan |
| 05:17.26 | drmessano | #OneAsteriskUnderSpencer |
| 05:18.21 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 05:45.31 | *** join/#asterisk Ahmadi (02b14c59@gateway/web/freenode/ip.2.177.76.89) |
| 05:45.37 | Ionic | Hmm... is a number without a national prefix the canonical way to represent numbers? This question sucks... |
| 05:45.39 | Ahmadi | Hi |
| 05:46.46 | Ionic | Basically, I have hooked up an ISDN card to my phone line and am using LCR and chan_lcr to process incoming and outgoing calls. The caller IDs I receive do not include the national prefix and I wonder if that's right |
| 05:47.13 | Ionic | It matches up with what mISDN sees, but doesn't make a lot of sense in my mind |
| 05:47.41 | WIMPy | That's the way the PSTN works. |
| 05:48.02 | WIMPy | Use LCRs screnn-in option to add the prefix. |
| 05:48.15 | Ionic | That doesn't work if I'm also using the msn config option... |
| 05:48.50 | WIMPy | On a line? |
| 05:49.32 | Ionic | I've hacked chan_lcr's code into submission and wrapped everything number-related I could find with a special function, but that sounds meh |
| 05:49.47 | Ionic | In general, you cannot use both the "msn" and "screen-in" options at the same time |
| 05:50.10 | WIMPy | No. Why would you? |
| 05:50.26 | Ahmadi | I have 2 Asterisk servers,(192.168.10.222 and 192.168.10.224), I have also a SIP Client that registered on 222, I'm trying to transfer a call with my SIP Client to asterisk 224 with "sip:200@192.168.10.224" uri. The problem is that the call transferred on extension 200 of current asterisk(222), not the asterisk 224 that i specified ip of it on the uri. |
| 05:50.31 | Ahmadi | Please help me |
| 05:51.06 | Ionic | Huh, why wouldn't I use both screen-in and specify MSNs to listen for? |
| 05:52.02 | WIMPy | There's no point in mangling prefixes if you match MSNs. |
| 05:52.32 | WIMPy | And What kind of interface are we talking about? |
| 05:53.14 | Ionic | Ahmadi: what function are you using to transfer the call? Might be helpful... |
| 05:53.28 | Ionic | WIMPy: was that question directed to me? |
| 05:53.41 | WIMPy | yes |
| 05:54.14 | Ionic | Hum... what interface do you mean exactly? The device? The driver? |
| 05:54.48 | WIMPy | You said you connected it to a line. But using MSN matching on a line doesn't make sense. |
| 05:55.04 | Ionic | To a BRI line |
| 05:56.02 | WIMPy | Then don't try to use the MSN parameter. |
| 05:56.56 | Ionic | So the workaround would be to specify screen-in and just not match MSNs I don't care about in asterisk's dialplan? |
| 05:57.50 | WIMPy | You could do it in Asterisks dialplan as well. But that's more work. |
| 05:58.12 | WIMPy | And you never want to match MSNs on a line. |
| 05:59.06 | Ionic | Why not? Currently, I want the device to not react to calls to specific MSNs, so using the "msn" option seemed like a good way to do this |
| 05:59.43 | WIMPy | That's not what it does. It limits caller IDs. |
| 06:00.03 | Ionic | Wait... what? |
| 06:00.24 | WIMPy | Calls to unconfigured numbers will be ignored anyway. |
| 06:02.52 | Ahmadi | Ionic, I'm transferring call with an third party component API, "call->transfer('sip:200@192.168.10.224')" |
| 06:03.35 | Ahmadi | I want know for transferring a call between 2 asterisk, is there any prerequisites that i must do? for example creating trunk/route or etc ?! |
| 06:03.44 | Ionic | Ahmadi: so uhm maybe that's broken or does not support transfering calls to other servers? |
| 06:06.06 | Ahmadi | I just need to know is any trunk/route/... need to defined on my first asterisk to transferring a call to another asterisk? |
| 06:06.22 | Ionic | Hum, yeah, looks like I can't filter out calls via the msn option... oookay... |
| 06:06.58 | Ionic | Ahmadi: to be honest I don't know, you'll have to wait for somebody who does :/ |
| 06:07.12 | Ahmadi | Ionic, the 3th party company say the API is correct and accept any URI for call transferring. |
| 06:07.24 | Ahmadi | ok |
| 06:07.33 | Ahmadi | Please somebody help me |
| 06:07.34 | WIMPy | If you want to filter in LCR, use dialing= or just use Asterisks Dialplan. |
| 06:11.20 | Ionic | ACK, I'd better filter it out in Asterisk, thanks |
| 06:11.41 | Ionic | Eventually I'll have to put the device into NT mode anyway |
| 06:12.07 | WIMPy | Or "filter" rather. As said, unconfigured numbers will ignored anyway. In both places. |
| 06:12.34 | WIMPy | That will be a different story then. |
| 06:17.36 | Ionic | Hmm... no, screen-in national % 0% doesn't seem to do anything, meh |
| 06:19.10 | Ionic | Probably because I'm bridging the interfaces |
| 06:20.03 | WIMPy | Hmm. I don't know if bridging ovverides that. I'm using the previous version with remote=asterisk. |
| 06:24.18 | Ionic | https://dpaste.de/C8Am/raw this config is routing everything to Asterisk and lets me also do outgoing calls via Asterisk |
| 06:24.22 | drmessano | Ahmadi: If your 3rd Party API accepts a SIP URI, and it's not following the URI when invoked, then the API is broken, the dev lied, or you're not implementing it properly |
| 06:25.00 | Ionic | Doesn't get a lot easier, but I suspect the bridge calls may interfere with the screeing |
| 06:27.48 | WIMPy | Well, I don't know the bridging thing. I prefer the old version as it allows me to easily do all routing from the Asterisk dialplan. |
| 06:28.51 | Ionic | I should probably test what happens if I don't bridge to asterisk both ways |
| 06:28.55 | Ahmadi | drmessano,But when the SIP Client pass the URI to the first Asterisk, How it detect the second Asterisk? maybe i need to define any in/out routes or any trunk between the two asterisks? |
| 06:29.10 | WIMPy | Sounds like a good idea. |
| 06:29.40 | Ionic | I've commented out the ast section completely and removed the bridge ast statement |
| 06:29.46 | WIMPy | But bridging is probably done really low level. So it's likely the issue. |
| 06:30.53 | Ionic | Now lcradmin state outputs the prefixed caller ID correctly (although there's a setup message probably from mISDN missing the leading zero) |
| 06:31.36 | WIMPy | That can't change. That's just what you receive. |
| 06:32.30 | Ionic | Yeah... hmmm |
| 06:32.51 | WIMPy | So just do an outdial to asterisk instead of bridging and you should be fine. |
| 06:33.07 | WIMPy | And in the other direction as well, off course. |
| 06:33.39 | Ionic | Via routing, I guess |
| 06:33.48 | WIMPy | Yes |
| 06:40.17 | Ionic | Hmm... now that's weird |
| 06:40.27 | Ionic | Ah, it's not |
| 06:42.36 | Ionic | With screening enabled, I see the "correct" phone number in LCR, but the non-prefixed one in Asterisk now |
| 06:42.49 | Ionic | Err, with screening DISABLED |
| 06:43.10 | WIMPy | o.O |
| 06:43.21 | Ionic | With screening enabled, I get a doubly-prefixed caller ID in LCR, but a one-time prefixed one in Asterisk |
| 06:43.30 | Ionic | Or maybe I'm reading LCR's messages wrong |
| 06:44.11 | WIMPy | Maybe the new asterisk interface has some built-in magic? Or maybe there's a config for it? |
| 06:44.58 | WIMPy | Or do you have screen-out on the asterisk interface? |
| 06:45.10 | Ionic | Moving screen-in to the ast interface didn't change anything |
| 06:45.23 | Ionic | Nope |
| 06:45.26 | WIMPy | No, because it's out there. |
| 06:46.12 | WIMPy | But if you add a prefix, you should also change the TON to unknown to avoid such issues. |
| 06:47.02 | WIMPy | screen-in national % unknown 0% |
| 06:47.07 | Ionic | hmmm, wait |
| 06:47.26 | WIMPy | Same for international. |
| 06:48.20 | Ionic | Huh, that syntax looks weird |
| 06:48.41 | Ionic | I have screen-in national % 0% and screen-in international % 00% |
| 06:54.05 | Ionic | Maybe I'm misinterpreting it, not sure |
| 06:54.06 | Ionic | https://dpaste.de/2VVf/raw |
| 06:54.23 | *** join/#asterisk MarkSX (~MarkSX@unaffiliated/marksx) |
| 06:54.32 | WIMPy | I can't look there. |
| 06:55.36 | drmessano | Ahmadi: Detect? Wut? It doesn't need to "detect" anything. You are literally passing it a SIP URI |
| 06:55.45 | Ionic | Why not? |
| 06:56.12 | WIMPy | Unsupported crypto. |
| 06:56.16 | Ionic | Oh |
| 06:58.37 | drmessano | Ahmadi: Your API isn't doing what you think it's doing. If you were simply using Asterisk Dial(), this is EVERYTHING Asterisk needs. You're literally giving it a user and a host. Nothing to "detect" or "be aware of". |
| 06:58.55 | Ionic | https://bpaste.net/raw/915b59ada576 maybe that works better |
| 06:59.22 | WIMPy | Nope. Same issue. |
| 06:59.24 | Ionic | drmessano: although Dial() won't "really" transfer the call itself |
| 06:59.44 | drmessano | Ionic: We're discussing context |
| 07:00.12 | drmessano | 'sip:200@192.168.10.224' <-- Enough |
| 07:00.15 | Ionic | Yep |
| 07:00.30 | Ionic | Even Transfer() should work that way |
| 07:00.37 | drmessano | You don't need to "tell" Asterisk what 192.168.10.224 is... |
| 07:01.08 | drmessano | or preconfigure it.. it's literally a full SIP URI pointing to another host |
| 07:01.18 | drmessano | So the API isn't doing what it says |
| 07:01.29 | drmessano | or it's being invoked improperly.. maybe syntax |
| 07:02.48 | Ionic | WIMPy: the gist essentially is this: "SETUP from CH(3) interface from=Ext caller id number=0<<<XYZ>>> ...", then screening kicks in "SCREEN (found in screen list) given type=national present=allowed id=<<<XYZ>>> used id=0<<<XYZ>>>" and then blackmagic changes it again "SETUP to CH(4) interface from=Ext caller id number=00<<<XYZ>>>" (all within LCR though, Asterisk seems to only ever see the "correc |
| 07:02.54 | Ionic | t" value of 0<<<XYZ>>>) |
| 07:03.32 | WIMPy | Did you add the "unknown"? |
| 07:03.38 | Ionic | Nope |
| 07:03.51 | WIMPy | Try that. |
| 07:05.08 | Ionic | Hum... yeah, that does the trick... but why? |
| 07:06.11 | Ionic | We're changing the type to unknown so that filterting only happens once, but... uhm.. |
| 07:06.25 | WIMPy | Beacuse you get a wrong number when you add a 0 and leave the TON to national. |
| 07:07.54 | Ionic | I'd expect further screening operations to not add anything if the number is already prefixed with the setup national or international prefix |
| 07:07.55 | Ahmadi | drmessano,ok |
| 07:07.57 | Ahmadi | thanks |
| 07:07.59 | Ionic | Oh well... |
| 07:08.44 | WIMPy | How would the next instance know you already added a prefix when the metadata still says it needs to be added? |
| 07:09.23 | Ionic | Comparing the beginning of the string |
| 07:09.40 | Ionic | I've seen LCR do this in nationalize_callerinfo() |
| 07:09.43 | WIMPy | That's not what you do. |
| 07:10.50 | Ionic | Well, that's actually removing the prefix if it detects any |
| 07:10.56 | WIMPy | It's the same as rotating JPEGs depending on the orientation bits without clearing them. |
| 07:38.02 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:59.54 | *** join/#asterisk Ahmadi (02b14b1f@gateway/web/freenode/ip.2.177.75.31) |
| 08:10.34 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 08:17.34 | *** join/#asterisk Tiffon (~name@unaffiliated/tiff0n) |
| 08:18.06 | Ahmadi | In following document, the author mention that each peer registration need username/password. but where can i define that username/password? |
| 08:18.06 | Ahmadi | http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html |
| 08:18.42 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 08:18.58 | Ahmadi | i mean "Notice that at the end of the registration line we tag on a forward slash and the username of the remote Asterisk box" line |
| 08:19.51 | [TK]D-Fender | That is not inherently a username |
| 08:20.06 | [TK]D-Fender | register => user:pass@host/regexten |
| 08:20.25 | [TK]D-Fender | that last setting only suggests what to call when dialplan BACK |
| 08:21.21 | Ahmadi | ok, But what is user:pass? |
| 08:21.26 | Ahmadi | where can i define it? |
| 08:21.34 | Ahmadi | its user/pass of asterisk root? |
| 08:21.41 | Ahmadi | or extension |
| 08:21.45 | Ahmadi | or ... |
| 08:21.51 | Ahmadi | [TK]D-Fender |
| 08:22.32 | [TK]D-Fender | You are registering to the other box |
| 08:22.44 | [TK]D-Fender | you have a PEER on that other box that you are registering to |
| 08:22.57 | [TK]D-Fender | that is where you registering against |
| 08:23.09 | [TK]D-Fender | [fred] |
| 08:23.14 | *** join/#asterisk _0x5eb_ (~seb@seb-hpws2.w1.tele.crt1.net) |
| 08:23.15 | [TK]D-Fender | defaultuser=thisistheusername |
| 08:23.26 | [TK]D-Fender | secret=thisisobviouslythepassword |
| 08:23.44 | *** join/#asterisk Jesterboxboy (~Thunderbi@213.147.167.20) |
| 08:24.02 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 08:25.13 | Ahmadi | [TK]D-Fender, I mean in the other box that is my another Asterisk, how can i define that user/pass? |
| 08:25.51 | UncleKiwi | in the /etc/asterisk/sip.conf file |
| 08:26.25 | UncleKiwi | you need to lear the concepts of 'friend' and 'peer' |
| 08:26.31 | UncleKiwi | *learn |
| 08:27.18 | *** join/#asterisk Blashyrkh (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 08:28.03 | [TK]D-Fender | [TK]D-Fender> you have a PEER on that other box that you are registering to <- |
| 08:28.08 | [TK]D-Fender | and I just GAVE you the sameple |
| 08:29.28 | UncleKiwi | sorry my messages were for Ahmadi |
| 08:30.30 | Ahmadi | ok |
| 08:30.56 | Ahmadi | I don't know different of "peer" , "friend" ;) |
| 08:31.23 | UncleKiwi | you are very new to asterisk right ? |
| 08:31.41 | Ahmadi | yes |
| 08:31.53 | UncleKiwi | some youtube clips are really good |
| 08:32.02 | UncleKiwi | but also |
| 08:32.04 | UncleKiwi | ~book |
| 08:32.05 | 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 |
| 08:33.48 | UncleKiwi | Ahmadi what is your intention > |
| 08:33.49 | UncleKiwi | ? |
| 08:34.13 | UncleKiwi | what is your end goal |
| 08:34.21 | Ahmadi | i want transfer call from Asterisk 1 to Asterisk 2. |
| 08:34.32 | Ahmadi | i installed both asterisks |
| 08:34.40 | UncleKiwi | seperate offices ? |
| 08:34.48 | Ahmadi | yes |
| 08:34.54 | UncleKiwi | ok |
| 08:35.02 | Ahmadi | but both are in same network |
| 08:35.17 | UncleKiwi | same building ? |
| 08:35.24 | UncleKiwi | same city ? |
| 08:35.29 | Ahmadi | Yes,Asterisk 1 is 192.168.10.222, Asterisk 2 is 192.168.10.224 |
| 08:35.48 | UncleKiwi | why do you need two asterisk boxes ? |
| 08:36.00 | [TK]D-Fender | peer matches by IP normally. user matches by username on the invite. Friend = user+peer |
| 08:36.00 | UncleKiwi | just curious |
| 08:36.13 | [TK]D-Fender | You can't Dial() a user, only friend or peer |
| 08:36.49 | [TK]D-Fender | typically if you don't have multiple hosts behind the same IP (typically due to NAT), then you can use PEER for everything else |
| 08:37.24 | Ahmadi | Im using AMI to manage event, i want split calls, because of some business goals |
| 08:37.36 | Ahmadi | UncleKiwi |
| 08:37.50 | UncleKiwi | ok sounds tricky |
| 08:38.01 | [TK]D-Fender | Your describing much more advanced goals when you don't understand the basics |
| 08:38.12 | [TK]D-Fender | this you're* |
| 08:38.23 | Ahmadi | [TK]D-Fender, ;) |
| 08:39.06 | Ahmadi | I just want do http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html |
| 08:39.26 | [TK]D-Fender | If those IP's are fixed you don't even need to bother registering. |
| 08:39.30 | Ahmadi | the problem is that the author considered that i know the basics, but i don't know |
| 08:39.35 | [TK]D-Fender | only dynamic hosts need to register |
| 08:39.54 | [TK]D-Fender | You're supposed to have gone through the book and actually registered softphones, etc |
| 08:40.10 | [TK]D-Fender | because that's no differen when setting * upt to talk to another * |
| 08:40.12 | [TK]D-Fender | SIP is SIP |
| 08:40.22 | [TK]D-Fender | the peer concept is the same |
| 08:41.03 | Ahmadi | Where trunk peer defined? in sip.conf? |
| 08:41.13 | [TK]D-Fender | yes |
| 08:41.30 | [TK]D-Fender | taht's the first thing the book tells you |
| 08:41.44 | Ahmadi | if yes, why when i define a trunk with FreePBX user interface , not exists in sip.conf? |
| 08:41.46 | [TK]D-Fender | If you want it to talk SIP... go make entries in sip.cofn |
| 08:42.10 | [TK]D-Fender | because you haven't learned how INCLUDE works |
| 08:42.19 | [TK]D-Fender | because it merges in ANOTHER file for those contents |
| 08:42.27 | Ahmadi | ok |
| 08:42.39 | Ahmadi | Do you know name of the included file? |
| 08:42.42 | [TK]D-Fender | And that's assuming you're using CHAN_SIP peers |
| 08:42.53 | [TK]D-Fender | you see it in the main <-------------- |
| 08:43.06 | [TK]D-Fender | include => obviousfilename here |
| 08:43.10 | Ahmadi | Is there any different between chan_sip and pjsip except of port number? |
| 08:43.16 | [TK]D-Fender | actually #include |
| 08:43.32 | [TK]D-Fender | yes, there is clearly a difference |
| 08:43.41 | [TK]D-Fender | the config files format does not look the same at all |
| 08:43.50 | WIMPy | But the port number is not one of them. |
| 08:43.54 | [TK]D-Fender | pjsip is more split up between section types |
| 08:44.16 | [TK]D-Fender | the port is different merely for the fact you can't have 2 things fighting over the same one |
| 08:44.26 | UncleKiwi | Hi WIMPy |
| 08:44.59 | *** join/#asterisk boris_t (~boris_t@363103629.convex.ru) |
| 08:45.33 | Ahmadi | thanks |
| 08:45.37 | Ahmadi | for the info |
| 08:46.14 | [TK]D-Fender | And you should not be touching the config files at all |
| 08:46.47 | Ahmadi | why? |
| 08:46.53 | *** join/#asterisk jkroon (~jkroon@uls-154-73-32-14.wall.uls.co.za) |
| 08:47.19 | [TK]D-Fender | because FreePBX OWNS them |
| 08:47.25 | [TK]D-Fender | You don't get to do things like this by hand |
| 08:47.39 | UncleKiwi | ahhhh he's running that |
| 08:47.41 | [TK]D-Fender | the GUI will TRASH whatever you think you're doing as soon as you apply a change |
| 08:47.54 | [TK]D-Fender | You have to play by its rules |
| 08:48.40 | Ahmadi | But the asterisk documents never teach solutions by FreePBX , always the documents say change files manually! |
| 08:49.21 | WIMPy | Sore. It's Asterisk documentation, not FreePBX documentation. |
| 08:49.26 | [TK]D-Fender | Yes and your car manual doesn't teach you how to drive like a sports car driver either |
| 08:49.41 | UncleKiwi | ahaha |
| 08:49.50 | [TK]D-Fender | Programming books don't tell you how to use MS Excel. Byut MS Excel is a PROGRAM, WHY NOT!>?!?!?!? |
| 08:50.16 | [TK]D-Fender | that GUI is **BOLT ON EXTRA** |
| 08:50.17 | Ahmadi | For example i want know how can i connect two asterisk pbx, but i can't search google for "how can i connect two freepbx" ? |
| 08:50.27 | [TK]D-Fender | You can't? |
| 08:50.35 | [TK]D-Fender | *IK* can google that and there are dozens of guides |
| 08:51.01 | [TK]D-Fender | If you can't google that and come up with specific guides then you must completely suck at Googling. |
| 08:51.15 | [TK]D-Fender | https://www.google.ca/#q=how+can+i+connect+two+freepbx |
| 08:51.21 | [TK]D-Fender | Hey look, RESULTS! |
| 08:51.32 | [TK]D-Fender | "Connecting Two FreePBX/Asterisk Systems Together Over the Internet ." |
| 08:51.38 | [TK]D-Fender | "Connecting two FreePBX machines together | FreePBX" |
| 08:51.48 | [TK]D-Fender | "IAX2 Trunk(s) between two (or more) FreePBX Servers - Tips and" |
| 08:51.59 | [TK]D-Fender | WOW, I wonder if one of THOSE might be what I'm looking for! |
| 08:52.09 | Ahmadi | ok, got it |
| 08:52.21 | UncleKiwi | Ahmadi if you don't learn the basic concepts of asterisk you are going to have a lot of confusing moments |
| 08:52.37 | [TK]D-Fender | Same with an inability to google. |
| 08:52.59 | [TK]D-Fender | And then more when you don't realize the rules your 3rd party GUI throws into this |
| 08:53.24 | UncleKiwi | i agree |
| 08:55.16 | UncleKiwi | Ahmadi do you have any of these asterisk boxes exposed to the internet ? |
| 08:55.25 | Ahmadi | nope |
| 08:55.32 | UncleKiwi | thats good |
| 08:55.34 | Ahmadi | both are in private network |
| 08:56.07 | UncleKiwi | things can get expensive when you dont take security seriously |
| 08:56.55 | UncleKiwi | just saying if you new take care |
| 08:57.29 | Ahmadi | thanks for the hint |
| 08:58.03 | UncleKiwi | yeah i know some people that lost 20K |
| 08:58.12 | UncleKiwi | and thats probably small |
| 09:08.50 | *** join/#asterisk tirej (~tirej@unaffiliated/tirej) |
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| 09:20.46 | [TK]D-Fender | heads off to bed |
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| 13:17.02 | Ahmadi | Hi |
| 13:17.13 | Ahmadi | Anyone can help me about following error: |
| 13:17.16 | Ahmadi | [2016-11-13 01:03:56] NOTICE[2048] res_pjsip_session.c: Call from 'AsteriskPj222' (UDP:192.168.10.222:5060) to extension '' rejected because extension not found in context 'from-pstn'. |
| 13:17.42 | Ahmadi | I have an incoming call, but rejected |
| 13:17.52 | Ahmadi | instead of passing to my extension |
| 13:18.05 | Ahmadi | i have a incoming route that must pass all calls to my extension |
| 13:36.43 | *** join/#asterisk Jesterboxboy (~Thunderbi@213.147.167.20) |
| 13:37.36 | Ahmadi | <PROTECTED> |
| 13:38.10 | Ahmadi | The above error appear if i have registration on my trunk, else the first error |
| 13:43.46 | *** join/#asterisk karelk (~karel@31.10.149.26) |
| 14:17.51 | *** join/#asterisk bidule_ (~bidule@APointe-a-Pitre-651-1-297-100.w90-31.abo.wanadoo.fr) |
| 14:18.25 | bidule_ | hello, I am having trouble to,set up an analog phone with spa3000, does anyone has experience with the spa3000 ? |
| 14:22.34 | *** join/#asterisk darkdrgn2k (~darkdrgn5@unaffiliated/darkdrgn2k) |
| 14:22.35 | darkdrgn2k | hi all |
| 14:22.45 | darkdrgn2k | i got a SPA-3102 |
| 14:22.52 | darkdrgn2k | but PSTN line reigstraion always says "failed" |
| 14:23.12 | darkdrgn2k | but i dont see an attempt at the asterisk side for the registrion? |
| 14:23.29 | darkdrgn2k | i can ping the ata from the asterisk server no problem |
| 14:26.03 | darkdrgn2k | syslog seems to come through |
| 14:26.08 | darkdrgn2k | but registration not so much |
| 14:26.33 | bidule_ | ok thank you but can you place call to pstn from line 1 ? |
| 14:27.04 | ttaylor | darkdrgn2k: what version of asterisk? sip.con or pjsip.conf? |
| 14:27.48 | darkdrgn2k | i think i can place a call |
| 14:28.09 | darkdrgn2k | just did a tcp dump and alhtough i see the registration being generated on the SPA i dont swee it in tcp=dump |
| 14:28.16 | darkdrgn2k | just the syslog entry |
| 14:29.01 | *** join/#asterisk Jesterboxboy (~Thunderbi@213.147.167.20) |
| 14:30.05 | darkdrgn2k | ov 12 09:29:06 REGISTER sip: 192.168.250.120 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.45.253:5061;branch=z9hG4bK-c0e81791#015#012From: 514222222251422222225142222222 <sip:5142222222@192.168.250.120>;tag=72ab8d6d5b96be0do1#015#012To: 5142222222 <sip:5142222222@192.168.250.120>#015#012Call-ID: 9138a253-3a773cde@192.168.45.253#015#012CSeq: 62362 REGISTER#015#012Max-Forwards: 70#015#012Contact: 5142222222 <sip:5142222222@192.168.45.253:5061>;e |
| 14:30.05 | darkdrgn2k | xpires=300#015#012User-Agent: Linksys/SPA3102-5.2.13(GW002)#015#012Content-Length: 0#015#012Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER#015#012Supported: x-sipura, replaces#015#012#015 |
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| 14:45.38 | darkdrgn2k | i dont even see it hitting the router |
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| 18:26.03 | drmessano | darkdrgn2k: Get any closer? |
| 18:27.35 | drmessano | darkdrgn2k: Make sure you're connected to the WAN/Internet port. The SIP client doesn't use the LAN port. |
| 19:00.36 | *** join/#asterisk qakhan (~qakhan@50-204-254-11-static.hfc.comcastbusiness.net) |
| 19:01.12 | qakhan | hi all, is there any open source VOIP monitoring tool |
| 19:21.07 | drmessano | What exactly do you want to monitor? |
| 19:21.44 | Mango45 | How does one log the X-RTP-Stat header, when the h extension runs before the phone sends that header? |
| 19:24.51 | qakhan | drmessano monitor the call quality. |
| 19:25.11 | drmessano | qakhan: Ok, and what you would you measure that in? |
| 19:25.13 | qakhan | how many calls are in progress etc... |
| 19:25.31 | drmessano | Ok, NOW we are getting somewhere |
| 19:25.39 | drmessano | You can do that with SNMP |
| 19:26.11 | drmessano | Pick a tool.. Set up SNMP with Asterisk.. done |
| 19:26.38 | qakhan | which tool |
| 19:26.53 | drmessano | Anything that monitors SNMP |
| 19:26.56 | WIMPy | Troule is that polling for number of channels doesn't really work. |
| 19:26.57 | drmessano | There are thousands |
| 19:27.16 | WIMPy | Unless you do it at a very high rate. |
| 19:27.33 | qakhan | how about monitor voice quality issue |
| 19:27.51 | drmessano | qakhan: Ok, and what metric do you want to monitor there? |
| 19:27.52 | WIMPy | Do you use voip phones? |
| 19:28.02 | drmessano | Network latency = Standard tools |
| 19:28.02 | qakhan | yes |
| 19:28.28 | WIMPy | Then the only place where you could monitor voice quality is on those phones. |
| 19:28.30 | *** join/#asterisk u0m3 (~u0m3@86.127.130.105) |
| 19:29.01 | WIMPy | Unless you only want to find some part of possible issues. |
| 19:29.59 | drmessano | qakhan: My point is no, there is no Microsoft VoIP Monitor 2016 R2 for monitoring "VoIP".. you monitor just as you would anything else.. SNMP for Asterisk statistics, monitor the health of the instance, look for network latency |
| 19:30.05 | qakhan | dont have access on those phones. they are on customer site |
| 19:30.21 | drmessano | Then theres nothing to measure for "quality" |
| 19:30.55 | drmessano | You can check the logs for registrations dropping if you have call completion issues.. that sort of thing |
| 19:31.13 | Mango45 | qakhan: FWIW, I'm trying to do the same thing. SIP has an X-RTP-Stat header which indicates lost packets, but I haven't figured out a way to get that. |
| 19:31.13 | WIMPy | You can take a loot at RTCP, but that can have its own issues. |
| 19:31.15 | drmessano | But thats all basic log monitoring tools |
| 19:33.15 | qakhan | Mango45 what is complete name of FWIW |
| 19:33.17 | qakhan | ? |
| 19:33.27 | drmessano | ..... |
| 19:33.29 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 19:33.31 | Mango45 | "For what it's worth" |
| 19:33.42 | drmessano | HE thinks you're giving him the name of a specific tool |
| 19:33.47 | drmessano | Which I keep telling him doesnt exist |
| 19:33.48 | Mango45 | Sorry about that. |
| 19:34.05 | drmessano | He wants a name of a VOIP TOOL he can google |
| 19:34.20 | drmessano | Im wasting my typing at this point |
| 19:34.41 | [TK]D-Fender | probably started a while ago :) |
| 19:34.54 | drmessano | [TK]D-Fender: You mean from the beginning? |
| 19:34.59 | WIMPy | Why don't you just name it? Like "wireshark"? |
| 19:35.04 | [TK]D-Fender | Yeah, probably somewhere around there... |
| 19:35.18 | drmessano | WIMPy: There you go |
| 19:35.30 | drmessano | qakhan: Use Wireshark.. It monitors everything |
| 19:35.54 | Mango45 | qakhan: They're being sarcastic. Wireshark is not a good tool for what you're after. |
| 19:36.02 | qakhan | i know wireshark but its just for run time |
| 19:36.32 | WIMPy | It is. You just have to collect the results manually. |
| 19:36.35 | Mango45 | He has a point though. If I get an audio quality complaint, I'd love to be able to look at CDRs and see how much packet loss was reported for that call. It would make troubleshooting a lot easier instead of just guessing. |
| 19:36.41 | drmessano | qakhan: Asterisk SNMP + Network monitoring .. Glue it all together with some NMS |
| 19:37.28 | qakhan | i need some tool which take save data |
| 19:37.38 | drmessano | ....... |
| 19:37.41 | qakhan | like network monitoring tools |
| 19:37.50 | drmessano | Thats what I just said |
| 19:38.01 | drmessano | 14:36:41 <drmessano> qakhan: Asterisk SNMP + Network monitoring .. Glue it all together with some NMS |
| 19:38.11 | qakhan | ok |
| 19:38.24 | drmessano | 1. Get an NMS of your choice |
| 19:38.29 | drmessano | 2. Configure Asterisk for SNMP |
| 19:38.35 | drmessano | 3. Monitor it and the network |
| 19:38.41 | drmessano | 4. ?????? |
| 19:38.44 | drmessano | 5. Profit |
| 19:39.21 | drmessano | If there's a firewall, set up SNMP there and monitor it too |
| 19:39.48 | drmessano | Thats how you monitor a VoIP |
| 19:40.04 | Mango45 | Is it possible to log SIP DEBUG to one file per channel? |
| 19:40.52 | [TK]D-Fender | no |
| 19:41.00 | [TK]D-Fender | not from * |
| 19:41.00 | Mango45 | Thanks. |
| 19:41.03 | qakhan | thanks drmessano |
| 19:41.15 | drmessano | Mango45: Thats the wrong approach anyway, IMHO.. Dump all the DEBUG to SYSLOG and get something that tears it down |
| 19:42.57 | drmessano | Powerful SYSLOG parsing > Flinging millions of organized tiny bits to disk |
| 19:51.12 | Samot | So if I'm understanding this right, he wants a tool that lets him parse the SIP headers in the packet? |
| 19:51.22 | Samot | So he can find one header that has information inside of it? |
| 19:51.56 | Samot | <PROTECTED> |
| 19:52.00 | Mango45 | Yes. |
| 19:52.10 | Samot | That's called OpenSER or something along that lines. |
| 19:52.10 | Mango45 | SIP_HEADER works, but only if I hang up first. If the far end hangs up first, H runs before my phone sends X-RTP-Stat |
| 19:52.20 | Mango45 | Righto. |
| 19:53.42 | WIMPy | YOu find RTP stats in the channel variables. But I wouldn't expect them to have much of a meaning. |
| 19:54.08 | Samot | What is the need of monitoring the RTP traffic? |
| 19:54.40 | Mango45 | So that if there's an audio quality complaint I can say whether or not it was caused by packet loss. |
| 19:55.31 | Samot | So you're going to monitor everything from both sides? |
| 19:56.16 | Mango45 | I'm just after the end-of-call stats. |
| 19:56.41 | Samot | OK so there's packet loss.. |
| 19:56.43 | Samot | Now what? |
| 19:56.55 | Mango45 | Fix the network. |
| 19:56.59 | Samot | If you're having packet loss on a call, you are probably having packet loss on data. |
| 19:57.16 | Samot | But where was the packet loss |
| 19:57.22 | Samot | Was it between your router and the ISP |
| 19:57.27 | Samot | Was it on your local network? |
| 19:57.40 | Mango45 | Not necessarily. I've dealt with ISPs that have poor routes specifically to my ITSP's media gateway, but good routes elsewhere. |
| 19:58.02 | Samot | Are you 100% your ITSP has a media gateway? |
| 20:00.52 | Samot | Generally that's a pretty easy question to answer. |
| 20:01.04 | Samot | There's like three options. Yes, No and I don't know. |
| 20:01.13 | Mango45 | Since you already said * can't do it, the question is kind of moot. |
| 20:02.01 | Samot | I've dealt with ISPs that have poor routes specifically to my ITSP's media gateway <--- How does that have to do with * |
| 20:02.36 | Samot | Knowing whether or not your ITSP has it's own media gateway has nothing to with with * |
| 20:03.03 | Mango45 | It doesn't matter whether the ITSP has a media gateway if I can't log the data I'm after. |
| 20:03.55 | Samot | Sure you can, you just need the right tools. Asterisk is not the right tool for it. |
| 20:04.45 | Samot | "Sorry, can't build your house. I was told using a pipe wrench to pound in nails wasn't the right tool. I don't have the energy to get the right tool to do the job" |
| 20:05.44 | *** join/#asterisk u0m3_ (~u0m3@5-12-97-145.residential.rdsnet.ro) |
| 20:09.44 | wyoung | so what's the issue? what can't asterisk do? |
| 20:10.23 | Mango45 | When the far end of a call hangs up first, the h extension runs before my phone sends the X-RTP-Stat header. Therefore, I can't access that header from the h extension. |
| 20:11.33 | Mango45 | "Sorry, can't install your baseboards. My hammer works on every type of nail in this house except baseboard nails. I have to get a different hammer just for this specific nail." |
| 20:12.25 | wyoung | Mango45: what does that header do again? |
| 20:12.47 | Mango45 | It tells me, among other things, if there was packet loss on the call. |
| 20:13.49 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 20:15.11 | *** join/#asterisk u0m3 (~u0m3@5-12-96-138.residential.rdsnet.ro) |
| 20:17.13 | *** join/#asterisk u0m3__ (~u0m3@188.25.10.191) |
| 20:18.29 | Mango45 | HAHAHAHAHA I GOT IT! |
| 20:20.02 | wyoung | so hang up first then |
| 20:20.18 | wyoung | wooooo! |
| 20:20.31 | Mango45 | haha, that'll be a great one to teach the users |
| 20:23.14 | wyoung | yup :) |
| 20:24.32 | Samot | Who's setting this header, BTW? |
| 20:24.39 | Mango45 | My phone |
| 20:25.01 | Samot | You know it's a non-standard header right? |
| 20:25.14 | Mango45 | why should I care? |
| 20:25.22 | *** join/#asterisk catphish (~J@unaffiliated/catphish) |
| 20:25.40 | Samot | Because not everything uses that header? |
| 20:25.53 | Samot | Cisco's version is P-RTP-Stats |
| 20:26.09 | Mango45 | I try to use as little Cisco as possible. |
| 20:26.19 | Samot | So don't assume that you can say "I have the SIP header X-Header" and expect people just to know about it. |
| 20:26.25 | Samot | When it's a non-standard header. |
| 20:26.48 | Samot | Using Cisco has nothing to do with it. |
| 20:27.04 | Samot | I was pointing out it's a header that is randomingly named and assigned. |
| 20:27.05 | catphish | would anyone possibly be able to point me in the direction of the code which matches a peer (specifically in the realtime cache) when a packet arrives in chan_sip? |
| 20:27.17 | Mango45 | IDGAF if it's non-standard, it has the data I want. |
| 20:27.40 | Samot | And missing the point. |
| 20:28.20 | Samot | catphish: What is the issue you are having? |
| 20:28.29 | Mango45 | I'M missing the point? Every time I come in here I wonder if I'm speaking a foreign language. |
| 20:29.23 | catphish | Samot: chan_sip seems to be making incoming requests to dynamic peers by IP in the realtime cache, i want it to only match static peers by IP |
| 20:29.25 | Mango45 | Samot: catphish wants to read the code which matches a peer (specifically in the realtime cache) when a packet arrives in chan_sip, and he wants to know where to find said code. |
| 20:30.05 | catphish | i'd like to read the code to see if this behaviour is intended, and if i can change it |
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| 20:32.08 | Samot | That's what permit is for. |
| 20:32.18 | catphish | permit? |
| 20:32.23 | Samot | deny/permit options in the peer settings. |
| 20:32.34 | Samot | deny=0.0.0.0/0.0.0.0 |
| 20:32.41 | Samot | permit=0.0.0.0/0.0.0.0 |
| 20:32.45 | Samot | 1) Deny from all |
| 20:32.57 | Samot | 2) Allow from all and since it's after deny, override deny. |
| 20:34.14 | catphish | i'm not sure how that would help |
| 20:34.22 | Samot | Well.. |
| 20:34.43 | Samot | You set permit to the IP or IP range you want to be allowed to access that peer for the endpoint. |
| 20:35.08 | Samot | So if you only want phones on 192.168.1.150/28 to be allowed, that's what you put in permit. |
| 20:35.34 | Samot | OR |
| 20:35.52 | Samot | You set host=192.168.1.150 for the device peer you only want to register from that IP |
| 20:37.10 | catphish | thnk you, but i don't think this is what i need |
| 20:37.38 | Samot | So you want to block the other incoming requests? |
| 20:37.52 | Samot | Only allow requests from certain IPs/hosts through? |
| 20:37.57 | Samot | And tell the rest to go to hell? |
| 20:38.24 | catphish | i want to prevent incoming SIP requests matching dynamically registered peers by IP address |
| 20:38.54 | Samot | Why would want to stop requests from registered endpoints? |
| 20:39.25 | catphish | because i have a static peer with the same IP address, and i want the requests to match *that* peer |
| 20:39.31 | Samot | When you say "dynamic" you mean the endpoint sending a REGISTER request? |
| 20:40.04 | Samot | You can't control where the requests come from. |
| 20:40.22 | Samot | You can deny or allow the requests based on where they come from. |
| 20:43.39 | catphish | i think i've found it: static int realtime_peer_by_addr(...) |
| 20:44.00 | catphish | and it appears that it *should* work the way i want, there's a comment /* First check for fixed IP hosts */ |
| 20:48.32 | *** join/#asterisk DelphiWorld (~VOIPER@openvpn/user/DelphiWorld) |
| 20:49.14 | DelphiWorld | hi people |
| 20:49.36 | DelphiWorld | i see asterisk 18 in openwrt. should that be asterisk 18, or 1.8? |
| 20:50.27 | catphish | must be 1.8, afaik the latest release is only 14 |
| 20:50.46 | DelphiWorld | hahaha so strange |
| 20:50.49 | DelphiWorld | that what i thought |
| 20:50.57 | DelphiWorld | what do we need for webrtc in asterisk 13? |
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| 22:25.17 | catphish | looks like there's a peers_by_ip table, which caches peers by ip regardless of whether they're dynamic or not |
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| 23:07.09 | masked | hey |
| 23:07.17 | masked | im having trouble with pjsip |
| 23:07.28 | masked | i try an outgoing call via and endpoint |
| 23:07.36 | masked | and it says it's an invalid uri |
| 23:11.55 | masked | [Nov 13 10:10:58] ERROR[100763]: res_pjsip.c:2883 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *, const char *, const char *): Could not create dialog to endpoint 'didlogic-trunk' as URI '61354424749' is not valid |
| 23:12.02 | masked | which causes no route to destination |
| 23:12.20 | masked | im no code wizard but it looks like it's failing for some coding specifics |
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| 23:21.06 | file | what is the configuration and the Dial string you are using? |
| 23:22.44 | masked | exten => _[1-9].,1,Dial(PJSIP/didlogic-trunk/${EXTEN}) |
| 23:22.55 | file | that format is not supported with PJSIP |
| 23:23.00 | masked | oh |
| 23:23.01 | file | it would be PJSIP/${EXTEN}@didlogic-trunk |
| 23:23.05 | masked | sweet |
| 23:23.07 | masked | thanks file |
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| 23:37.26 | masked | file: that helped. now i'm getting a 404 no destination |
| 23:37.46 | masked | i think it's because my ua's go through opensips |
| 23:37.50 | masked | and asterisk is behind it |
| 23:38.10 | masked | and asterisk is sending requests out to the upstream directly instead of going through opensips |
| 23:39.04 | masked | what would be ideal is if the request was sent out and the itsp spoke back directly to the UA |
| 23:39.33 | masked | i guess that would be re-invite like behaviour |
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| 23:40.09 | masked | and here i was thinking i wanted to implement cgrates on opensips and keep the signalling go through opensips, but perhaps i just want it on the media gateways |
| 23:40.36 | masked | either or really if i can hand off the media |
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