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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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07:51.44 | duo_ubuntu | hi, can someone help me with ChanSpy please? |
08:12.35 | duo_ubuntu | my dialplan is simple https://pastebin.com/wzxQnmFZ, can someone help me to make it work? |
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17:28.13 | duo_ubuntu | Does ChanSpy not include in asterisk 11? |
17:30.19 | duo_ubuntu | ok, have it |
17:30.20 | duo_ubuntu | sorry |
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17:50.09 | Samot | What is the issue you are having? The dialplan example you posted earlier had no chanspy commands in it. |
18:25.15 | Samot | duo_ubuntu: ^^^^ |
18:25.32 | duo_ubuntu | hi Samot, |
18:26.52 | duo_ubuntu | so far I just add this : exten=> 191,2. ChanSpy(SIP/GSM/${EXTEN}) |
18:28.21 | Samot | Yeah, you can't do that |
18:28.32 | Samot | ${EXTEN} is the extension you are in. |
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18:28.38 | Samot | You're telling it to spy on 191 |
18:29.19 | duo_ubuntu | not work also with : exten=> _*222x.#,n,ChanSpy(SIP/GSM/${EXTEN:4},q) |
18:29.57 | Samot | No # |
18:30.02 | duo_ubuntu | we put it on extensions.conf right? |
18:30.08 | Samot | Yes |
18:30.34 | Samot | Also it's X not x |
18:31.12 | Samot | _*222X. |
18:31.58 | duo_ubuntu | its not working when I call *2228591 , 8591 is the extension who is on the line |
18:32.23 | ac2 | Hi, perhaps someone here can help. Are the Cisco 79XX series phones capable of using TLS and SRTP with an asterisk pbx server? |
18:32.43 | ac2 | (7942, 7962, 7975) |
18:33.20 | duo_ubuntu | or how to make all the extension with XXXX like my dialplan example can be monitor with chanspy? |
18:33.21 | Samot | exten => _*222X.,1,ChanSpy(SIP/GSM/${EXTEN:4},q) |
18:35.01 | duo_ubuntu | let me try, but how to make my all dialplan for _XXXX as my example can be monitor? |
18:37.31 | duo_ubuntu | so any call still goin on, how I can monitor with just press *222, without need to know the extension number? |
18:39.28 | duo_ubuntu | Samot, I try to call *2225237, 5237 is the extension, but there is no sound |
18:39.40 | Samot | Uhm |
18:40.11 | Samot | Because you have q |
18:40.42 | duo_ubuntu | maybe on the extensions.conf need to make [chanspy] before extension? |
18:40.49 | duo_ubuntu | let me try erase q |
18:41.21 | duo_ubuntu | call is disconnect without q |
18:42.31 | Samot | Have you not read how ChanSpy works? |
18:42.33 | duo_ubuntu | it said, unable to open beep , no such file |
18:44.34 | duo_ubuntu | I 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? |
18:44.44 | Samot | exten => *222,1,ChanSpy(SIP/GSM) <-- That by itself will look for any channel that begins with SIP/GSM |
18:45.16 | Samot | Well it will look at the SIP with the peer GSM |
18:45.48 | Samot | Once ChanSpy is running you can dial 5237# and it will jump to that peer |
18:45.56 | Samot | Probably you're going to have is your peer is GSM |
18:46.21 | Samot | You're trying to spy on outbound calls, correct? |
18:46.52 | duo_ubuntu | any call made and bridging all I want |
18:47.08 | Samot | OK then you just want ChanSpy(SIP) |
18:47.46 | duo_ubuntu | try *222 and call is ended :-( |
18:47.49 | Samot | More specifically, ChanSpy(SIP,b) |
18:48.04 | duo_ubuntu | any idea what happen? |
18:48.06 | Samot | exten => *222,1,ChanSpy(SIP,b) |
18:48.11 | Samot | I have no idea. |
18:48.20 | Samot | You need to show debugs if you want those types of answers. |
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18:50.00 | duo_ubuntu | it wrote on asterisk log, unable to openbeep (format gsm): no such file or dir |
18:50.55 | Samot | Did you make the changes I just gave? |
18:51.18 | Samot | You cannot do ChanSpy(SIP/GSM/${EXTEN} |
18:51.20 | Samot | You cannot do ChanSpy(SIP/GSM/${EXTEN}) |
18:51.27 | Samot | Because GSM is the peer |
18:51.46 | Samot | You can do ChanSpy(SIP/GSM) but you want *all* bridged calls... |
18:51.48 | duo_ubuntu | exten => *222,1,ChanSpy(SIP/GSM) |
18:51.56 | Samot | What is GSM? |
18:52.00 | Samot | Is that your trunk? |
18:53.05 | duo_ubuntu | its was a user on sip-custom-contexts.conf |
18:53.23 | Samot | Is it an actual peer? |
18:53.44 | duo_ubuntu | yes |
18:54.33 | Samot | Pastebin the log file output for a failed attempt |
18:54.44 | duo_ubuntu | ok wait |
18:55.01 | duo_ubuntu | I need to login in that laptop, wait a sec |
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18:57.07 | duo_kali | Samot, here https://pastebin.com/rL8zy2d9 |
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18:58.22 | Samot | Your missing the beep file. |
18:58.30 | Samot | Did you not install all the sound files? |
18:58.51 | Samot | Make it ChanSpy(SIP/GSM,q) and do it again. |
18:58.54 | Samot | So the output |
18:59.10 | duo_kali | mmm I think it was install automatically? or we need to install manual? |
18:59.22 | duo_kali | ok wait |
18:59.40 | Samot | You have to tell it during the install to install the sound files and which ones to use |
19:00.44 | duo_kali | ahhh its working now |
19:01.55 | duo_kali | thanks Samot |
19:03.12 | duo_kali | Samot, is there a way... the call is made is dial to my extension to monitor? |
19:03.22 | duo_kali | is there a way to do that? |
19:03.34 | Samot | The same way |
19:03.48 | Samot | You can either do ChanSpy(SIP/yourext) |
19:04.10 | Samot | Or ChanSpy(SIP) then dial <exten># to jump to your channel |
19:04.11 | duo_kali | so we dont need to call, but chanspy call automatic to my extension |
19:04.40 | Samot | Sure, you can originate a call with a local channel to your extension |
19:04.51 | Samot | When you answer it fires off ChanSpy for you |
19:06.25 | duo_kali | exten => #,1,ChanSpy(SIP/5555,q) ? I still not get it |
19:06.27 | duo_kali | :-( |
19:06.50 | Samot | Get what? |
19:07.12 | duo_kali | how to make the extension for what I want :-) |
19:07.20 | Samot | I just told you. |
19:08.12 | Samot | When 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. |
19:08.30 | Samot | You are going to have to earn some dialplan and basics about Originate and Local channels. |
19:08.43 | Samot | s/learn/earn |
19:09.11 | duo_kali | yes it is. |
19:11.11 | duo_kali | thanks Samot for your help. it mean alot to me :) |
19:11.39 | duo_kali | I will learn more about more basics and originate and local chanels |
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23:13.36 | sicelo | i 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-Fender | Show us the complete call debug with SIp debug enabled |
23:38.26 | [TK]D-Fender | !pb |
23:38.31 | [TK]D-Fender | ~pb |
23:38.31 | infobot | i 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 | ^^^ |
23:42.56 | sicelo | http://paste.debian.net/1043876/ |
23:43.22 | sicelo | that's basically what scrolls through. |
23:43.30 | [TK]D-Fender | Not at all what we should be looking at |
23:43.33 | [TK]D-Fender | no core debug |
23:43.36 | [TK]D-Fender | verbose 10 |
23:43.38 | [TK]D-Fender | no rtp debug |
23:43.49 | [TK]D-Fender | sip debug enabled |
23:44.00 | [TK]D-Fender | and nothing masked |
23:44.42 | sicelo | not 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 |
23:52.32 | sicelo | http://paste.debian.net/1043878/ |
23:55.10 | [TK]D-Fender | <[TK]D-Fender> "core set verbose 10" <------------------------ |
23:55.22 | [TK]D-Fender | nvm |
23:55.24 | [TK]D-Fender | missed something |
23:56.30 | [TK]D-Fender | Peer audio RTP is at port 41.184.234.227:7078 |
23:56.37 | [TK]D-Fender | That's our key issue |
23:56.58 | [TK]D-Fender | c=IN IP4 41.184.234.227 |
23:57.06 | [TK]D-Fender | Which it's trusting from your peer. |
23:57.32 | [TK]D-Fender | the 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" |
23:58.45 | sicelo | let me try (with comedia/force_rport), since yes is deprecated |
23:59.19 | sicelo | i 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-Fender | not global |