IRC log for #asterisk on 20190330

00:01.17Samothttps://tools.ietf.org/html/rfc3966
00:38.01life_of_eInteresting, does asterisk even do anything with it?  The default settings on the ATA has it disabled so it's not needed for normal SIP functions
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02:14.26life_of_ewhere are custom sounds supposed to live on the system?  I currently have them in /usr/share/asterisk/sounds/custom next to the standard set but I can't get them working from the dialplan when I do a playback(custom/filename)
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02:39.19life_of_enm found it
02:56.10wyounggg
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14:53.07wonderworldhey, i want to create an account for the asterisk forums https://community.asterisk.org/. when clicking "Sign Up", i am just presented with a login dialog. where would i sign up?
14:55.24filehttps://signup.asterisk.org/
15:13.38wonderworld<PROTECTED>
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16:59.30life_of_eI'm trying to figure out how to transfer a call into ConfBridge but I don't have it quite right.  I've got a feature code for blind transfer configured (#1) which does ask for the extension.  I have an extension set up as exten => 5000,1,ConfBridge(1).  Transferring to x5000 just yielded a busy signal
17:00.15life_of_eThe Dial() commands for the endpoints all use options t and T
17:04.29[TK]D-FenderIt's not in the right condition or place
17:04.41[TK]D-FenderShow your actual dialplan and the attempt
17:09.48life_of_ehttps://pastebin.com/zNDZZyvd
17:10.34[TK]D-FenderYou're using an SPA SIP phone?
17:10.45[TK]D-FenderCisco/Linksys?
17:10.51life_of_eThat was the calling phone but yes
17:11.07[TK]D-Fenderok, what are you transfering from?
17:11.15life_of_eAn analog phone on the ATA
17:12.08life_of_eI wanted to get transfers working on the analog phones.  Transfers work fine on the SPAs
17:12.31[TK]D-FenderAlmost all ATA's support stanard transfer functions using the hook-flash
17:12.53[TK]D-FendersDTMF features for that are only for the most limited of devices
17:12.57life_of_eYeah, but same result, as soon as I gave it the extension I created to join the ConfBridge I got a busy signal
17:13.11[TK]D-Fenderon the ATA?
17:13.17life_of_eyes
17:13.33[TK]D-Fenderbut the SPA seems to be in the confbridge, right?
17:13.42[TK]D-Fenderwhich I see executing
17:13.46life_of_eI had channel audio, did the transfer, then got busy signal.  It appeared to be but when I dialed the bridge no one was in it.
17:14.44life_of_eI'll have to play with it some more.  As far as I can tell it should work.
17:15.14life_of_eThe busy signal is the oddball
17:15.34[TK]D-Fenderpost transfer that is not a "busy" disgnal
17:15.37[TK]D-Fenderthat is a REORDER tone
17:15.52[TK]D-Fenderbasically the telco saying "you're done, hang up"
17:16.33[TK]D-Fenderanalog phones don't have a "logical" hangup the way a SIP phone does which is independent of you hanging up the handset onto the hookswitch
17:16.35life_of_eThen the ATA is confused about what the reorder tone should sound like.  It's giving me the slow busy
17:16.55[TK]D-FenderMaybe you haven't got it set to the right region settings
17:17.39[TK]D-FenderIs see your SPA sat in that conf for 13 seconds before leaving
17:18.02[TK]D-Fenderthens tarted a new call to the confbridge
17:18.41[TK]D-Fenderso you ware the SPA... leaving the confibridge... after being transfered TO it... and then calling back in and being the only one there...
17:18.42life_of_eRight, the SPA showed it was in use (red lamp) and I dialed into the bridge from the ATA
17:18.51[TK]D-Fenderthe ATA never called in... so who were you expecting to find?
17:19.55life_of_eThe SPA went on hold, I hit the line button to bring back the active line and it dropped the connection
17:20.00[TK]D-Fender9421 called in.  was answered by HT.  HT transferred 9421.  9421 was in the conf bridge.  9421 LEFT.  9421 called into the conf bridge DIRECTLY and is again the ONLY one there.
17:20.08[TK]D-FenderThe HT was NEVER shown to have called into the bridge
17:20.27[TK]D-Fenderyour procedule looks broken
17:20.30[TK]D-Fenderprocedure
17:21.06life_of_eYeah, seems to be.  I'll play around with it some more.  But the region settings on the ATA are correct and the definitions for the reorder and busy tones are different.  It just played the busy tone not the reorder tone.
17:21.19[TK]D-Fenderok, that is a separate issue
17:21.39life_of_eI know it is, it was just odd
17:22.01life_of_eI expected either reorder or even a new dialtone
17:25.56life_of_eJust working out final details of the various configurations so I can switch over from the direct line connection
17:26.30life_of_eIf it plays busy, ok, I'll just make a note of that in the cheat sheet
18:02.05life_of_eIt works now, the problem wasn't where I thought.  Dirty switch contact in my test phone keypad causing multiple DTMF tones
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19:41.05Max13Hello there, I'm still new to Asterisk, and I'm experiencing an issue with IVR and DTMF. I have set an announcement when a call is received, then it's transferred to an IVR, which also plays an announcement and allows caller to enter an extension, with timeout set to 3 (3 seconds I guessed). When I enter a valid and reachable extension during the timeout (after the announcement is played), logs show "User enters [correct extension]" but consi
19:41.05Max13dered invalid. If I enter the extension DURING the announcement, it's recognized as valid and the extension rings. Am I missing something here ?
19:41.57[TK]D-FenderShow us the actual call
19:41.59[TK]D-Fender~pb
19:41.59infobothmm... 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
19:42.01[TK]D-Fender^^^
19:42.35Max13Do you need DTMF debug as well?
19:45.55[TK]D-FenderShouldn't
19:45.58[TK]D-Fendershow both calls
19:46.04Max13Ok
19:46.16[TK]D-FenderWe'll see after the first round
19:48.22Max13First call: https://pastebin.com/yANuSPL3
19:49.45Max13I'm calling 99933 which is a custom extension sending me to simulate incoming call from a specific trunk. The actual trunk number is redacted (+33700000000), and I try to call the same extension (999 typing 999)
19:50.08Max13This is when I wait the end of the IVR announce to type 999
19:52.40Max13This is when I enter 999 DURING the IVR announcement: https://pastebin.com/7kY2EjEb . And my extension rings
19:52.42[TK]D-FenderWhat "custom extension" is this?
19:52.53[TK]D-Fendernvm... diodn't read that properly
19:55.11[TK]D-FenderIs this a stock FreePBX IVR menu?
19:55.41Max13If by "stock" you mean created by it, yes
19:55.59Max13I didn't manually edit any file
19:56.10[TK]D-Fenderit looks very poorly designed
19:56.16[TK]D-FenderAre you on the latest version?
19:56.22[TK]D-FenderAll modules updateed?
19:56.38Max13Asterisk 13 on Raspberry Pi (raspberry-asterisk)
19:56.49[TK]D-FenderFREEPBX version
19:56.52Max13All modules updated through the Freepbx gui
19:56.55Max13I chekc
19:57.13[TK]D-Fender"dialplan show ivr-2" <- pb
19:57.24Max1314.0.5.25
19:58.25Max13https://pastebin.com/1ATftCzX
20:01.06[TK]D-Fender"dialplan show from-did-direct-ivr" <- pb
20:01.56Max13https://pastebin.com/0qa3vCdj
20:05.25[TK]D-Fenderok, something looks buggy here...
20:06.11[TK]D-FenderWe should add a diagnostic line to extension_additional.conf to validate a few things...
20:06.25Max13What can I do?
20:06.30[TK]D-Fender1 sec
20:08.03[TK]D-Fenderexten => s,n,NoOp(readstatus = ${READSTATUS} value = ${IVREXT})
20:08.23[TK]D-Fendergo search for [ivr-2], look under "s" and add it before this line:
20:08.31[TK]D-Fender<PROTECTED>
20:08.40[TK]D-FenderWhich indicates the 13th line of "s"
20:09.04[TK]D-Fenderpastebin that section of the dialplan when you have done this and we'll validate then apply it directly
20:09.09[TK]D-Fenderfor a sanity check
20:09.13Max13ok
20:13.10Max13https://pastebin.com/hmitSXbh
20:14.26[TK]D-Fenderlooks about right.  do "dialplan reload and retest where you wait before entiring the #
20:14.30[TK]D-Fenderwhere it failed before...
20:14.43Max13ok
20:14.51[TK]D-FenderI'm thinking the app isn't returning what the dialplan is designed thinking it should.
20:14.54Max13Thx, I check
20:17.33Max13https://pastebin.com/tesjJdma
20:23.32[TK]D-Fender[2019-03-30 21:15:48] VERBOSE[19324][C-00000021] pbx.c: Executing [s@ivr-2:13] NoOp("PJSIP/999-0000001c", "readstatus = TIMEOUT value = 999") in new stack
20:23.45[TK]D-Fenderok, crazy idea... do 999#
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20:26.16Max13Sorry, internet issue
20:26.20[TK]D-Fender<[TK]D-Fender> [2019-03-30 21:15:48] VERBOSE[19324][C-00000021] pbx.c: Executing [s@ivr-2:13] NoOp("PJSIP/999-0000001c", "readstatus = TIMEOUT value = 999") in new stack
20:26.21[TK]D-Fender<[TK]D-Fender> ok, crazy idea... do 999#
20:26.39Max13Why not xD
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20:27.22Max13Well...
20:27.51Max13My extension rings SO IMMEDIATELY, it's even faster than when I don't wait the end of the announcment
20:28.00Max13+e
20:28.17Max13So yes, 999# works
20:29.04[TK]D-Fenderthis is afreepbx IVR bug
20:30.25Max13Let me check if during the announcement, 999# is also recognized
20:31.23Max13It doesn't
20:31.44[TK]D-Fenderthe entire way they design their IVR looks wasteful....
20:31.47lostinsiphello
20:32.02Max13If it's a FreePBX IVR bug, does it mean there is a line (or many) in [avr-2] that needs to be removed?
20:32.07Max13Or added ?
20:32.08lostinsipI have a weird problem I hope someone may have some idea about
20:32.24lostinsipI run a FreePBX system, and that is all working fine
20:32.33[TK]D-FenderMax13, they way they use Read() is flawed.  Read REQUIRES "#" not to be considered a timeout.
20:32.45[TK]D-Fenderexten => s,n,GotoIf($["${READSTATUS}" = "TIMEOUT" & "${IVREXT}" = ""]?t,1)
20:32.46lostinsipthe issue is related to VLAN tag implemnetation on the endpoints im using
20:32.57[TK]D-Fenderthis line doesn't see it as a "failure" because there is a partial value
20:33.23[TK]D-Fenderexten => s,n,GotoIf($["${READSTATUS}" != "OK"]?i,1) <- but if it isn't OK then apparently having a value isn't good enough so it never consdiers it at all
20:33.29[TK]D-Fenderthis line is the problem
20:35.08Max13I will report it to FreePBX then. Is there a workaround? Or do I have to change the announcement saying to end with "#" :p ?
20:36.40[TK]D-FenderFreePBX dialplan is wiped out and regenerated so it's tricky to circumvent.
20:37.11[TK]D-FenderThis could be making a copy of the IVR, correcting it, and using a Custom Destination to lead to that IVR.
20:37.21[TK]D-Fenderor hacking the module code so it builds it right
20:37.49Max13For now I'm just "playing", so I will continue playing, thank you
20:38.59Max13What I can do is play "type an extension followed by #" as an announcement, then a bip, then IVR without any announce, this way, you have to wait the "bip" and the end of announcement
20:40.03Max13Anyway, thanks a lot for you help and your time
20:40.59[TK]D-FenderYou're welcome.  Please post this issue up on their tracker so it can be addressed.  This is really not a good thing for it to be generating
20:41.20Max13I will send you the issue #
20:41.27[TK]D-FenderNo need for me
20:41.31Max13Ok then
20:41.33Max13/bbl
20:41.41[TK]D-FenderI am not on any of the dev teams
20:41.56Max13I meant, if you want to keep track :p
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