IRC log for #asterisk on 20180923

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02:33.44*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0-rc3 (2018/09/20), Standard: 15.6.1 (2018/09/20); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
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07:51.44duo_ubuntuhi, can someone help me with ChanSpy please?
08:12.35duo_ubuntumy dialplan is simple https://pastebin.com/wzxQnmFZ, can someone help me to make it work?
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17:28.13duo_ubuntuDoes ChanSpy not include in asterisk 11?
17:30.19duo_ubuntuok, have it
17:30.20duo_ubuntusorry
17:44.03*** join/#asterisk sarthor (~sarthor@unaffiliated/sarthor)
17:50.09SamotWhat is the issue you are having? The dialplan example you posted earlier had no chanspy commands in it.
18:25.15Samotduo_ubuntu: ^^^^
18:25.32duo_ubuntuhi Samot,
18:26.52duo_ubuntuso far I just add this : exten=> 191,2. ChanSpy(SIP/GSM/${EXTEN})
18:28.21SamotYeah, you can't do that
18:28.32Samot${EXTEN} is the extension you are in.
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18:28.38SamotYou're telling it to spy on 191
18:29.19duo_ubuntunot work also with : exten=> _*222x.#,n,ChanSpy(SIP/GSM/${EXTEN:4},q)
18:29.57SamotNo #
18:30.02duo_ubuntuwe  put it on extensions.conf right?
18:30.08SamotYes
18:30.34SamotAlso it's X not x
18:31.12Samot_*222X.
18:31.58duo_ubuntuits not working when I call *2228591 , 8591 is the extension who is on the line
18:32.23ac2Hi, perhaps someone here can help.  Are the Cisco 79XX series phones capable of using TLS and SRTP with an asterisk pbx server?
18:32.43ac2(7942, 7962, 7975)
18:33.20duo_ubuntuor how to make all the extension with XXXX like my dialplan example can be monitor with chanspy?
18:33.21Samotexten => _*222X.,1,ChanSpy(SIP/GSM/${EXTEN:4},q)
18:35.01duo_ubuntulet me try, but how to make my all dialplan for _XXXX as my example can be monitor?
18:37.31duo_ubuntuso any call still goin on, how I can monitor with just press *222, without need to know the extension number?
18:39.28duo_ubuntuSamot, I try to call *2225237, 5237 is the extension, but there is no sound
18:39.40SamotUhm
18:40.11SamotBecause you have q
18:40.42duo_ubuntumaybe on the extensions.conf need to make [chanspy] before extension?
18:40.49duo_ubuntulet me try erase q
18:41.21duo_ubuntucall is disconnect without q
18:42.31SamotHave you not read how ChanSpy works?
18:42.33duo_ubuntuit said, unable to open beep , no such file
18:44.34duo_ubuntuI try to read, when we wan to spy or listen the conversation between 2 line, we need to call it and listen on our phone?
18:44.44Samotexten => *222,1,ChanSpy(SIP/GSM) <-- That by itself will look for any channel that begins with SIP/GSM
18:45.16SamotWell it will look at the SIP with the peer GSM
18:45.48SamotOnce ChanSpy is running you can dial 5237# and it will jump to that peer
18:45.56SamotProbably you're going to have is your peer is GSM
18:46.21SamotYou're trying to spy on outbound calls, correct?
18:46.52duo_ubuntuany call made and bridging all I want
18:47.08SamotOK then you just want ChanSpy(SIP)
18:47.46duo_ubuntutry *222 and call is ended :-(
18:47.49SamotMore specifically, ChanSpy(SIP,b)
18:48.04duo_ubuntuany idea what happen?
18:48.06Samotexten => *222,1,ChanSpy(SIP,b)
18:48.11SamotI have no idea.
18:48.20SamotYou need to show debugs if you want those types of answers.
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18:50.00duo_ubuntuit wrote on asterisk log, unable to openbeep (format gsm): no such file or dir
18:50.55SamotDid you make the changes I just gave?
18:51.18SamotYou cannot do ChanSpy(SIP/GSM/${EXTEN}
18:51.20SamotYou cannot do ChanSpy(SIP/GSM/${EXTEN})
18:51.27SamotBecause GSM is the peer
18:51.46SamotYou can do ChanSpy(SIP/GSM) but you want *all* bridged calls...
18:51.48duo_ubuntuexten => *222,1,ChanSpy(SIP/GSM)
18:51.56SamotWhat is GSM?
18:52.00SamotIs that your trunk?
18:53.05duo_ubuntuits was a user on sip-custom-contexts.conf
18:53.23SamotIs it an actual peer?
18:53.44duo_ubuntuyes
18:54.33SamotPastebin the log file output for a failed attempt
18:54.44duo_ubuntuok wait
18:55.01duo_ubuntuI need to login in that laptop, wait a sec
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18:57.07duo_kaliSamot, here https://pastebin.com/rL8zy2d9
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18:58.22SamotYour missing the beep file.
18:58.30SamotDid you not install all the sound files?
18:58.51SamotMake it ChanSpy(SIP/GSM,q) and do it again.
18:58.54SamotSo the output
18:59.10duo_kalimmm I think it was install automatically? or we need to install manual?
18:59.22duo_kaliok wait
18:59.40SamotYou have to tell it during the install to install the sound files and which ones to use
19:00.44duo_kaliahhh its working now
19:01.55duo_kalithanks Samot
19:03.12duo_kaliSamot, is there a way... the call is made is dial to my extension to monitor?
19:03.22duo_kaliis there a way to do that?
19:03.34SamotThe same way
19:03.48SamotYou can either do ChanSpy(SIP/yourext)
19:04.10SamotOr ChanSpy(SIP) then dial <exten># to jump to your channel
19:04.11duo_kaliso we dont need to call, but chanspy call automatic to my extension
19:04.40SamotSure, you can originate a call with a local channel to your extension
19:04.51SamotWhen you answer it fires off ChanSpy for you
19:06.25duo_kaliexten => #,1,ChanSpy(SIP/5555,q) ? I still not get it
19:06.27duo_kali:-(
19:06.50SamotGet what?
19:07.12duo_kalihow to make the extension for what I want :-)
19:07.20SamotI just told you.
19:08.12SamotWhen user A makes a call, you need to also originate a call to *your* extension via a Local channel that will execute ChanSpy when you answer.
19:08.30SamotYou are going to have to earn some dialplan and basics about Originate and Local channels.
19:08.43Samots/learn/earn
19:09.11duo_kaliyes it is.
19:11.11duo_kalithanks Samot for your help. it mean alot to me :)
19:11.39duo_kaliI will learn more about more basics and originate and local chanels
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23:13.36siceloi have a problem with NAT, and would appreciate some help.  my asterisk server is behind NAT, and i have a SIP client on the same localnet as the asterisk server. the client has no audio when calling the asterisk server to play test sounds, e.g. hello-world. rtp set debug on shows packets with the external IP address available on the network. i am not doing port forwarding on the router because this asterisk installation is meant to be i
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23:38.23[TK]D-FenderShow us the complete call debug with SIp debug enabled
23:38.26[TK]D-Fender!pb
23:38.31[TK]D-Fender~pb
23:38.31infoboti guess 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.
23:38.33[TK]D-Fender^^^
23:42.56sicelohttp://paste.debian.net/1043876/
23:43.22sicelothat's basically what scrolls through.
23:43.30[TK]D-FenderNot at all what we should be looking at
23:43.33[TK]D-Fenderno core debug
23:43.36[TK]D-Fenderverbose 10
23:43.38[TK]D-Fenderno rtp debug
23:43.49[TK]D-Fendersip debug enabled
23:44.00[TK]D-Fenderand nothing masked
23:44.42sicelonot sure i follow
23:46.54[TK]D-Fender"core set debug off"
23:46.58[TK]D-Fender"core set verbose 10"
23:47.15[TK]D-Fender"sip set debug on" <- if using chan_sip
23:52.32sicelohttp://paste.debian.net/1043878/
23:55.10[TK]D-Fender<[TK]D-Fender> "core set verbose 10" <------------------------
23:55.22[TK]D-Fendernvm
23:55.24[TK]D-Fendermissed something
23:56.30[TK]D-FenderPeer audio RTP is at port 41.184.234.227:7078
23:56.37[TK]D-FenderThat's our key issue
23:56.58[TK]D-Fenderc=IN IP4 41.184.234.227
23:57.06[TK]D-FenderWhich it's trusting from your peer.
23:57.32[TK]D-Fenderthe peer seems to assuing yoru WAN IP it's behind, so you need to tell your peer to ignore the offer
23:57.51[TK]D-Fender"nat=yes" <- for "6001"
23:58.45sicelolet me try (with comedia/force_rport), since yes is deprecated
23:59.19siceloi had put those in global, and it worked somewhat, but for the first 3 seconds or so, it would use public IP, and switch
23:59.57[TK]D-Fendernot global

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