| 00:00.01 | *** join/#asterisk bkruse (~Adium@162.17.78.101) |
| 00:01.19 | xochilpili | [TK]D-Fender, http://picpaste.com/Captura_de_pantalla_2015-07-09_a_la_s__19.00.13-ZT3ubYqv.png |
| 00:02.05 | WIMPy | And then it's not even Asterisk but FreePBX. |
| 00:02.29 | [TK]D-Fender | There is a pjsip specific screen and you need to tell it your EXTENSION is NAT'd |
| 00:03.49 | *** join/#asterisk BBone_1 (~Thunderbi@s75-157-233-55.bc.hsia.telus.net) |
| 00:05.01 | xochilpili | [TK]D-Fender, in extensions settings there is not that nat'd option |
| 00:05.09 | Cyford33 | maybe he is using direct media |
| 00:05.11 | xochilpili | it's in sip settings? |
| 00:05.22 | [TK]D-Fender | EXTENSION |
| 00:05.31 | [TK]D-Fender | that should have settings for BOTH of these things |
| 00:07.07 | xochilpili | [TK]D-Fender, im in the extension '9001' and in it, there's that option; should i have to edit pjsip.conf to put it nat=yes;? |
| 00:07.19 | xochilpili | there's not that option |
| 00:07.23 | [TK]D-Fender | Show us |
| 00:10.13 | xochilpili | http://picpaste.com/Captura_de_pantalla_2015-07-09_a_la_s__19.08.09-TU7u0dX8.png |
| 00:10.39 | xochilpili | http://picpaste.com/Captura_de_pantalla_2015-07-09_a_la_s__19.08.11-fWRnYUfD.png |
| 00:11.20 | xochilpili | http://picpaste.com/Captura_de_pantalla_2015-07-09_a_la_s__19.08.37-Fvdqu1kX.png |
| 00:11.44 | xochilpili | http://picpaste.com/Captura_de_pantalla_2015-07-09_a_la_s__19.08.50-xH52SzbF.png |
| 00:12.05 | WIMPy | Dial options Ttr? That looks really evil. |
| 00:12.17 | xochilpili | http://picpaste.com/Captura_de_pantalla_2015-07-09_a_la_s__19.08.59-9XUwAZ2n.png |
| 00:12.41 | *** join/#asterisk Cyford33 (~cyford@c-73-207-183-115.hsd1.ga.comcast.net) |
| 00:14.53 | xochilpili | http://picpaste.com/Captura_de_pantalla_2015-07-09_a_la_s__19.09.07-aV5Zs59H.png |
| 00:15.25 | [TK]D-Fender | Media USe & the next 4 settings |
| 00:15.36 | [TK]D-Fender | check these, and then check your global PJSIP settings page |
| 00:22.27 | xochilpili | http://picpaste.com/Captura_de_pantalla_2015-07-09_a_la_s__19.21.49-2HuKft7G.png |
| 00:25.05 | [TK]D-Fender | What about those blanks at the bottom? |
| 00:27.00 | xochilpili | that's supossed to be in general settings |
| 00:27.59 | xochilpili | http://picpaste.com/Captura_de_pantalla_2015-07-09_a_la_s__19.27.32-eHCnTxx4.png |
| 00:28.50 | [TK]D-Fender | Ok, I can't confirm what PJSIP is doing wwith this |
| 00:29.06 | [TK]D-Fender | Unless you need specific functionality it offers I highly recommend for you to switch to chan_sip |
| 00:29.44 | xochilpili | oks |
| 00:29.53 | xochilpili | im going to use a sip extension |
| 00:30.20 | [TK]D-Fender | that will be on a different port if you do. |
| 00:30.25 | [TK]D-Fender | more forwarding to set up |
| 00:30.45 | [TK]D-Fender | And you'll have to change for this on your client |
| 00:36.48 | xochilpili | port 5061, right? |
| 00:38.08 | *** join/#asterisk hfp (~hfp@65.93.186.95) |
| 00:38.34 | xochilpili | i got unauthorized 401 |
| 00:40.20 | [TK]D-Fender | Ok, I can't spend an entire evening having to walk trough every single step and ask for proper backup every time. |
| 00:40.36 | [TK]D-Fender | punches his card for this ticket. |
| 00:43.36 | Cyford33 | when my dialplan trys to dial an unavailable sip trunk why do i get an Answer dialstatus... right after getting this error "Unable to request channel SIP/FreeServer" |
| 00:43.54 | *** join/#asterisk theron (~theron@2620:10d:c0a1:1110:8000::218b) |
| 00:43.57 | Cyford33 | http://pastebin.com/QZbXgCW9 |
| 00:45.08 | [TK]D-Fender | You're in "h" |
| 00:45.12 | [TK]D-Fender | that call has already ENDED |
| 00:45.13 | WIMPy | That is an Originate, i.e. another call. |
| 00:45.44 | [TK]D-Fender | And I'm not seeig a CDR to indicate you're referring to the channel you're attempting to originate |
| 00:45.56 | Cyford33 | yes, status is success sorry not answer |
| 00:46.09 | [TK]D-Fender | Show us... |
| 00:46.26 | [TK]D-Fender | we have no matching data yet and a partial dead call. |
| 00:47.06 | [TK]D-Fender | And we have no idea what you're doing in that actual verbose we see. |
| 00:48.33 | Cyford33 | i am originting a call, if it fails it goes to a differnt provider, problem is it returns a success when it clearlly fails |
| 00:48.41 | Cyford33 | callbacks |
| 00:49.17 | [TK]D-Fender | I don't see what anything returns yet |
| 00:49.24 | [TK]D-Fender | I don't know what is actually being called in there |
| 00:49.31 | [TK]D-Fender | show us the actual dialplan, the bad CDR's, etc |
| 00:49.44 | [TK]D-Fender | Right now I have no proof of 2/3 of this scenario |
| 00:50.29 | WIMPy | And we are still on a hung up call. |
| 00:51.06 | Cyford33 | yes, i dialback after a hangup |
| 00:51.12 | [TK]D-Fender | Also a really bad place to start... |
| 00:51.15 | [TK]D-Fender | .... |
| 00:51.22 | [TK]D-Fender | Cyford33, show us the dialplan |
| 00:51.35 | [TK]D-Fender | is pretty sure he knows what it is right now.... |
| 00:51.50 | Cyford33 | http://pastebin.com/jw0f53M4 |
| 00:51.52 | [TK]D-Fender | And the call |
| 00:52.02 | [TK]D-Fender | the full call |
| 00:52.24 | Cyford33 | ok 1 sec. it takes everyone down |
| 00:52.34 | [TK]D-Fender | you have the logs |
| 00:52.42 | [TK]D-Fender | just show us the rest of THAT call. |
| 00:52.47 | [TK]D-Fender | Showing shouldn't take anythig down |
| 00:54.23 | Cyford33 | <PROTECTED> |
| 00:54.41 | Cyford33 | ill reproduce it, 1 min |
| 00:54.54 | [TK]D-Fender | hold on. |
| 00:54.57 | [TK]D-Fender | might not need |
| 00:55.10 | WIMPy | And you should start using lables instead of explicit priorities. |
| 00:56.52 | Cyford33 | http://pastebin.com/xk2HzWi1 |
| 00:56.57 | [TK]D-Fender | Looks like it failed on 2 retries (not sure why it even tried twice) |
| 00:57.15 | [TK]D-Fender | But still returned "success" |
| 00:58.19 | [TK]D-Fender | Which doesn't look sane |
| 00:58.24 | Cyford33 | if it really is successfull it should return Answer right? |
| 00:58.32 | [TK]D-Fender | No. |
| 00:58.38 | [TK]D-Fender | that is not what the apps instructions say |
| 00:59.07 | Cyford33 | ok, thats what motif channel returns |
| 00:59.35 | Cyford33 | originate an dial do seem to return differntlly though |
| 01:00.01 | [TK]D-Fender | Motif channels don't "return" anything |
| 01:00.03 | Cyford33 | ohh dial returns answer lol |
| 01:00.26 | [TK]D-Fender | <Cyford33> when my dialplan trys to dial an unavailable sip trunk why do i get an Answer dialstatus... right after getting this error "Unable to request channel SIP/FreeServer" |
| 01:00.29 | Cyford33 | well asterisk returns a status when dialing out motif |
| 01:00.49 | [TK]D-Fender | And no, we AREN'T looking at "dialstatus" |
| 01:01.01 | [TK]D-Fender | DIAL returns a DIALSTATUS |
| 01:01.07 | [TK]D-Fender | DIAL != ORIGINATE |
| 01:01.07 | Cyford33 | yes |
| 01:01.15 | [TK]D-Fender | Stop mising the terms and values up |
| 01:01.19 | Cyford33 | correct, thts what i was sayin |
| 01:01.27 | [TK]D-Fender | ok, well this looks like a bug so far |
| 01:01.32 | Cyford33 | i said there NOT alike |
| 01:01.36 | Cyford33 | i use both |
| 01:01.38 | [TK]D-Fender | I'd recommend you change your approach on this |
| 01:01.44 | Cyford33 | this is originate |
| 01:01.59 | Cyford33 | ok |
| 01:02.10 | Cyford33 | only 1 trunk does this |
| 01:02.28 | Cyford33 | and if it fails while online it works as expected |
| 01:02.34 | [TK]D-Fender | use a call file or CLI originate and use it to start a LOCAL CHANNEL. In ther do the dial YORUSELF and check the status in the dialplan to handle the failover you're trying to do in "h" |
| 01:02.56 | [TK]D-Fender | What ver of * is this exactly? |
| 01:03.29 | Cyford33 | 11.16 |
| 01:05.00 | [TK]D-Fender | update immediately |
| 01:05.04 | [TK]D-Fender | Then validate |
| 01:05.23 | [TK]D-Fender | then either file a bug report, and/or try the other approach I gave you |
| 01:05.32 | xochilpili | why always try to load pjsip to register instead of chan_sip if i have a sip extension? |
| 01:07.03 | [TK]D-Fender | xochilpili, because you're targeting the wrong port |
| 01:07.12 | [TK]D-Fender | xochilpili, I told you you'd have to change your CLIENT to match |
| 01:07.18 | [TK]D-Fender | xochilpili, And forward that other port |
| 01:07.28 | [TK]D-Fender | but you are asking for PAIN by bothering to use both in the first palce |
| 01:07.41 | [TK]D-Fender | Save yourself the trouble and jsut convert back to chan_ssip globally |
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| 01:20.56 | Cyford33 | i found the cuase |
| 01:21.26 | Cyford33 | it only does that when i check "Check to always try next trunk" in freepbx lol |
| 01:21.55 | Cyford33 | uncheck it my dialplan works as expected and goes to another trunk |
| 01:22.49 | Cyford33 | status HANGUP now when other box offline |
| 01:24.24 | Cyford33 | Continue if Busy: Check to always try next trunk ---- was the setting full setting what ever that actually does |
| 01:27.11 | [TK]D-Fender | um... makes no sense |
| 01:27.18 | [TK]D-Fender | those are DIALPLAN decisions, and this is your custom code |
| 01:27.22 | [TK]D-Fender | they do not mix |
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| 01:28.44 | Cyford33 | maybe freepbx trys to use a diffrnt trunk an got an answer, then my diaplan thought the call was successful and sent it to hangup |
| 01:33.04 | [TK]D-Fender | no |
| 01:33.14 | [TK]D-Fender | there is no "trunk" YOU are dialing stuff direct |
| 01:33.38 | [TK]D-Fender | Where do you see FreePBX dialplan doing anything "trunk" in there? |
| 01:33.41 | [TK]D-Fender | This is YOUR code |
| 01:33.58 | WIMPy | Maybe there's another interesting h extension? |
| 01:36.50 | Cyford33 | yes |
| 01:37.18 | Cyford33 | <PROTECTED> |
| 01:37.27 | [TK]D-Fender | Because this has nothing to do with it |
| 01:37.32 | [TK]D-Fender | And you shoudl know this already |
| 01:37.41 | Cyford33 | well i check it an it dont work |
| 01:37.49 | [TK]D-Fender | this is 1 fairly evident bug with that ver of * |
| 01:37.58 | [TK]D-Fender | upgrade, validate, then do as I told you |
| 01:37.58 | Cyford33 | <PROTECTED> |
| 01:38.05 | [TK]D-Fender | No, it doesn't |
| 01:38.19 | [TK]D-Fender | that is not a Dial |
| 01:38.50 | [TK]D-Fender | You've alrady confirmed that if the peer fails for ANOTHER reason it reacts right |
| 01:38.52 | Cyford33 | but freepbx failover would dial |
| 01:38.54 | [TK]D-Fender | this is NOT a guessing game |
| 01:39.06 | [TK]D-Fender | You've already confirmed how it reacts in the other case |
| 01:39.09 | [TK]D-Fender | This is a clear BUG |
| 01:39.16 | [TK]D-Fender | Now go prove if it has alredy been fixed |
| 01:39.20 | [TK]D-Fender | And then deal with it |
| 01:39.48 | Cyford33 | lol i dont need freepbx to try failover, since i do it myself |
| 01:39.53 | Cyford33 | <PROTECTED> |
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| 02:12.26 | Micc | If I make a call to some random extension like 01 and it's a menu or something, can I set the caller id that will show on the calling phone? |
| 02:13.52 | [TK]D-Fender | It's your dialplan |
| 02:13.55 | [TK]D-Fender | Do whatever you want |
| 02:14.41 | Micc | Yeah but how do I do that? HINTNAME? |
| 02:15.00 | Micc | CALLERID(dnid) ? |
| 02:15.07 | [TK]D-Fender | "core show function CONNECTEDLINE" |
| 02:16.08 | Micc | I think that's it. Thanks. |
| 02:18.15 | Micc | Tested and works exactly as expected. :) |
| 02:23.07 | [TK]D-Fender | You're welcome |
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| 02:53.43 | Micc | I'm registered for Astricon for the first time. I've been at this over 10 years and have never been. |
| 02:56.06 | [TK]D-Fender | same here |
| 02:56.17 | [TK]D-Fender | mut I'll wager the trip would be harder for me. |
| 02:56.19 | [TK]D-Fender | but* |
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| 02:57.35 | xochilpili | [TK]D-Fender, i just want to thank you again |
| 02:57.42 | xochilpili | [TK]D-Fender, now is working |
| 02:58.09 | xochilpili | and im sorry about all the headeaks that i make you pass |
| 02:58.29 | xochilpili | the issue was fixed with a sip trunk instead od pjsip trunk |
| 02:58.34 | xochilpili | instead of |
| 02:59.56 | xochilpili | headache |
| 03:00.37 | Micc | TK, why is it harder for you? |
| 03:01.47 | [TK]D-Fender | Do you have to cross an international border? |
| 03:02.04 | Micc | No, I suppose that would make it harder. |
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| 10:04.16 | tobiP | I need help regarding parkandannounce |
| 10:04.33 | tobiP | [Jun 23 15:11:11] VERBOSE[22537][C-00013bd4] features.c: == Parked SIP/xxxxxx-0000d29b on 2140000001 (lot default). Will timeout back to extension [default] s, 1 in 124 seconds |
| 10:04.38 | tobiP | [Jun 23 15:11:11] VERBOSE[22537][C-00013bd4] pbx.c: -- Added extension '2140000001' priority 1 to parkedcalls |
| 10:04.43 | tobiP | [Jun 23 15:11:11] VERBOSE[22537][C-00013bd4] app_parkandannounce.c: -- Call parked in space: 2140000001, timeout: 124000, return-context: default,s,1 |
| 10:04.48 | tobiP | [Jun 23 15:11:11] WARNING[22537][C-00013bd4] channel.c: No channel type registered for 'Console' |
| 10:04.53 | tobiP | [Jun 23 15:11:11] NOTICE[22537][C-00013bd4] channel.c: Unable to request channel Console/dsp |
| 10:04.57 | tobiP | [Jun 23 15:11:11] WARNING[22537][C-00013bd4] app_parkandannounce.c: PARK: Unable to allocate announce channel. |
| 10:05.06 | tobiP | [Jun 23 15:11:11] VERBOSE[22537][C-00013bd4] pbx.c: == Spawn extension (default, s, 1) exited non-zero on 'Parked/SIP/xxxxxx-0000d29b<ZOMBIE>' |
| 10:05.15 | WIMPy | ~pb |
| 10:05.15 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 10:05.18 | tobiP | what is happening here? |
| 10:06.09 | tobiP | http://pastebin.ca/3055461 |
| 10:06.48 | tobiP | sorry didn't meant to flood the channel |
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| 10:24.59 | phix | WIMPy, WIMPy, WIMPy!!!! |
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| 10:25.17 | WIMPy | Oi! |
| 10:25.34 | phix | :D |
| 10:26.41 | racho | hello, can someone shed some light on why do i get this -> "TLS failure occurred on RTP instance '0x7fe11c00ec68' due to reason 'no shared cipher'" on incomming calls to a websocket client while the outgoing ones work perfectly |
| 10:26.48 | racho | asterisk 11.18.0 |
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| 12:59.55 | andoh | For all those who answered me: I figured out the problem and it was totally unrelated to my agi script. It turned out that when we hired a new dba, they futzed with some things and it broke other things. So, after fixing those things my agi script worked fine. |
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| 15:11.40 | Dovid | I need a test call made to a German free phone number from a mobile phone. Anyone here that can make a quick test call? |
| 15:13.56 | Cyford33 | cant call from ya cell? |
| 15:16.30 | WIMPy | Number? |
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| 15:17.57 | Dovid | WIMPY: 08006644875 |
| 15:22.51 | WIMPy | It asks for a PIN or #. |
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| 15:39.44 | monsterco | can a user disable migrated directory from voicemail options? |
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| 16:01.00 | rim` | Does anyone have any experience getting a phone to work on a network that has commercial phone service? As in the DSL company already provides phone service to a residence so the phone can make outbound calls but the asterisk server shows the phone/extension as unreachable and calling that extension goes straight to voicemail. |
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| 16:02.15 | WIMPy | "The phone"? What phone? "The Asterisk Server"? What Asterisk server? |
| 16:02.36 | monsterco | rim - maybe change the port from 5060 to like 9060 |
| 16:02.36 | cmendes0101 | Yah was about to just say ???? |
| 16:02.40 | monsterco | probably conflict of ports |
| 16:02.42 | WIMPy | And where does the provider come in to play? |
| 16:03.07 | monsterco | provider has phone service enabled on the DSL modem; this is happening more and more nowadays... |
| 16:03.56 | WIMPy | That has been the standard for over a decade. But we have no idea what the question is actually about. |
| 16:04.02 | cmendes0101 | the customers have voip phones thats served off the dsl? |
| 16:04.43 | rim` | Yes, thanks, monsterco. Yes cmendes0101. I'm trying to get an asterisk phone to work on a network that already has phone service provided by the DSL company. |
| 16:04.49 | WIMPy | Not the modems or rather IADs have FXS and BRI ports. |
| 16:05.08 | WIMPy | What is an Asterisk phone? |
| 16:05.17 | cmendes0101 | WIMPy: Oh, gotcha |
| 16:05.27 | WIMPy | Your description is extremely vague. Where is the phone located and where the Asterisk? |
| 16:05.28 | monsterco | rim` - check asterisk sip debugs; your packets are most likely staying at DSL modem and not beng forwarded to voip phone ; use different port |
| 16:05.57 | rim` | monsterco, got it, thank you. |
| 16:06.55 | monsterco | rim` - you might also have an option to turn off voip on the DSL modem if it's not being used. Some other settings on modems like that effect SIP as well, like ALG etc...I hate those modems... |
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| 16:07.47 | WIMPy | Luckily the VOIP service oftn doesn't use the same internet connection ih which case you're safe. |
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| 16:46.34 | Milenco | hello again |
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| 16:47.02 | Milenco | I'm working on moving away from Macro's and into GoSub but I've got the Dial cmd parameter M to trigger a Macro on connect |
| 16:47.31 | [TK]D-Fender | So change them |
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| 16:47.40 | Milenco | can I replace that with a GotoIf(Dialstatus=ANSWER) line after the the Dial command on do I need to do it somewhat else? |
| 16:48.57 | [TK]D-Fender | Milenco: M() is run BEFORE bridging. your other idea is only after the call has ENDED and only if the call even continues processing |
| 16:49.06 | [TK]D-Fender | Milenco: Thse 2 things are nothing at all alike |
| 16:49.20 | Milenco | hmm, figured it might be something like that |
| 16:50.13 | Milenco | can i do something similar with GoSub then? To initiate a GoSub command before bridging? |
| 16:52.08 | [TK]D-Fender | Milenco: You should consider reading Dial()'s instructions..... |
| 16:52.28 | Milenco | hahah |
| 16:52.29 | Milenco | im sorry :D |
| 16:52.43 | Milenco | so..just use U |
| 16:53.14 | [TK]D-Fender | Milenco: Not so hard, was it? |
| 16:53.54 | Milenco | haha indeed |
| 16:54.02 | Milenco | i feel so stupid now :D |
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| 17:40.23 | Demon_VoIP | asterisk 13.4 PJSIP endpoint status "Invalid". What's wrong? |
| 17:42.07 | Demon_VoIP | OPTIONS got answer: SIP/2.0 200 Got it |
| 17:48.14 | [TK]D-Fender | we don't see proper status dumps and have no idea what that is a response to. We don't see enough to comment on. |
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| 19:58.30 | pjensen00 | I'm trying to use the ARI function "redirect". Previously I've only done the redirects to different channels on the same asterisk server. We have our own SIP server that directs traffic, and I want to be able to redirect the call back to that server. I have an endpoint set up in PJSIP.conf but whenever I try to put 'endpoint-1' in the 'endpoint' field for the ARI command 'redirect' it just tells me the resource isn't found. I can pu |
| 19:58.30 | pjensen00 | ll taht endpoint back when I query for a list of endpoints though. Any ideas? |
| 20:00.54 | rrittgarn | i havent used ARI, but do you have to specify technology when doing a redirect in ARI? like PJSIP/endpoint-1 ? |
| 20:02.25 | pjensen00 | Hrm, that may be. |
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| 20:02.40 | pjensen00 | The documentation claimed it had to be the same tech, so I thought it was implied. |
| 20:02.50 | pjensen00 | No harm in trying, here we gooooo! |
| 20:04.03 | pjensen00 | no dice |
| 20:07.50 | pjensen00 | pulling back the endpoint list it says that the state is 'online' |
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| 20:25.38 | pjensen00 | mjordan: I'm trying to use the redirect function in ARI. I'm using 'endpoint-1' as the endpoint varible, and even though I can pull it back through the GET endpoints with a state of 'online', the redirect 404's on me. :( |
| 20:26.54 | pjensen00 | Is there something I'm missing about the endpoint even though it's supposedly 'online'? |
| 20:27.15 | pjensen00 | Both the originating channel and the endpoint are pjsip |
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| 20:35.09 | mjordan | pjensen00: It should just be PJSIP/{endpoint_name} to redirect a channel to that endpoint, keeping in mind that it's either going to send a 302 to the inbound channel with a location of that PJSIP endpoint's AoR's contact, or a REFER request to the same |
| 20:35.21 | mjordan | generally, you should get some WARNING message on the CLI if that doesn't occur |
| 20:35.40 | mjordan | otherwise, I'd need to see your pjsip config, your actual REST API request, and the output from the CLI when that happens |
| 20:43.12 | pjensen00 | I've tried variations including PJSIP/endpoint-1 but I still get the 404. :( |
| 20:43.40 | pjensen00 | I did find your ticket and read it through though! |
| 20:44.59 | pjensen00 | Alright, I'll pastebin my current pjsip.conf (scrounging) |
| 20:47.58 | pjensen00 | pjsip.conf with IPs redacted (super spy stuff!) http://pastebin.com/dFgxhCDy |
| 20:54.17 | pjensen00 | and the response was http://pastebin.com/jVg2QdvX |
| 20:59.03 | file | redirect isn't in 13.2.0 |
| 20:59.07 | file | it was added in 13.3.0 |
| 20:59.12 | pjensen00 | ..... oh |
| 20:59.21 | pjensen00 | Well that explains that. |
| 20:59.22 | pjensen00 | :| |
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| 21:14.48 | mjordan | Yay new features! |
| 21:18.59 | pjensen00 | Upgrade complete, let's do this! |
| 21:20.31 | pjensen00 | success! I have no idea where the call went, but it got redirected somewhere! |
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