IRC log for #asterisk on 20170620

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00:20.10*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); 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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01:31.23jeffspeffhttps://www.youtube.com/watch?v=44uYz6PuTj0
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09:41.53ofg|brtrhello people
09:42.02[sID]hi
09:42.52ofg|brtrI need to start a SIP server. I don't have any telephony hardware. What I have is a Virtual server with a valid IP.
09:43.01ofg|brtrHow should I go about it.
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09:43.27ofg|brtrIt would be very helpful if someone enlighten  me.
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09:55.00ofg|brtr[sID]: ^^
10:03.50[sID]Do you recommend to use chan_pjsip? Is it better to sit on chan_sip ?
10:04.37[sID]ofg|brtr: look freepbx.org
10:06.13ofg|brtr[sID]: So I can setup a sip with freepbx.
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11:48.24pawieckiAnyone has some experience with KWS 6000?
11:55.28pawieckihmm, nevermind, bad question. It looks like after importing config backup, KWS reverted to default settings, with red led flashing.
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12:16.53pawieckiprobably interrupted backup file, because another one worked. Anyway not fun.
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14:00.09ziz212Dear All, I am having problem writing bash AMI script with telnet
14:00.26*** join/#asterisk skywayskase (~skywayska@163.182.162.226)
14:03.54pawieckiziz212: tell us WHAT is the problem - describe it.
14:04.12Samotziz212: As [TK]D-Fender said in the other channel, AMI is not a language.
14:04.39SamotSo you're writing a bash script that telnet's to Asterisk to issue an AMI command...
14:04.56ziz212Yes exactly write.
14:05.10SamotAnd the problem is?
14:05.26ziz212after the telnet session AMI commands are not running
14:05.37ziz212script stop from there
14:05.49SamotShow the script
14:05.50Samot~pb
14:05.58infobotsomebody said pastebin was 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.
14:05.58ziz212It wait for Stdin
14:06.01SamotAre you sending any?
14:06.44ziz212Action: Login Username: testuser Secret: testsecret
14:06.48SamotOK
14:06.51SamotThat logs you in.
14:07.05ziz212It never execute after the telnet command
14:07.10ziz212it waits for stdin
14:07.16SamotRight
14:07.19ziz212not taking from the script next line
14:07.25SamotShow the script
14:07.27Samot~pb
14:07.28infobotit has been said that 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.
14:07.48pawieckiI'm used to naming sip.conf accounts with their extension phone numbers, but lately i've been thinking about using mac address as the sip account name (also suggested in * book). The only two disadvantages i see are: 1. not familiar for me, and 2. still requires reprovisioning of the phone, to change display name / label. Do you guys just go with extension or use mac for sip account name?
14:08.32SamotNever have I seen it where the MAC is the username.
14:08.53SamotIn regards to the "sip account" for the device.
14:09.04ziz212OK. http://paste.lisp.org/display/349140
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14:11.33[TK]D-Fender<ziz212> It wait for Stdin <- no
14:11.46[TK]D-FenderAMI does NOT use stdin/sdtout
14:12.00[TK]D-FenderAMI is a TCP SOCKET CONNECTION
14:13.20ziz212Once I execute this, it wait me to input command on the shell. I can write commands and get the response. It never go beyond the telnet line
14:13.53SamotI don't use telnet with AMI but I'm guessing your cat should include all of it
14:13.57SamotIn one
14:13.59SamotNot three.
14:15.31SamotBut I could be wrong.
14:15.40ziz212I am trying to add featuremap as shell script
14:15.53SamotOK.
14:16.28ziz212I need to blind transfer the call to meeting room once originator press dtmf sequence .
14:16.52SamotLet's get past making the commands run first.
14:17.03ziz212I edited the features.conf file and add the line. but I dont know how to do the blind transfer to that
14:17.04SamotFigure out how to hammer in the nail...
14:17.06SamotThen build the house.
14:17.07ziz212it is soo dificult
14:17.22[TK]D-Fender<ziz212> I edited the features.conf file and add the line. but I dont know how to do the blind transfer to that <- you don't transfer to a feature
14:17.46[TK]D-FenderYou don't seem to understand any of the pieces you are using...
14:18.05[TK]D-Fenderfeatures EXECUTE on the channel they are bound to
14:18.16[TK]D-Fenderthere is no TRANSFER
14:18.23[TK]D-FenderYou call whatever you have it call
14:18.23ziz212I think I am wasting my time ...
14:18.31[TK]D-FenderI think you don't understand the words you're using
14:18.32SamotIf you're originating a call to someone...
14:18.37SamotAnd they press 22...
14:18.49SamotYou aren't transferring to them to conf bridge...
14:18.57SamotYou are _sending_ them to it
14:19.11SamotYou can put them right in it.
14:19.37ziz212Please let me have some word for you. Freepbx comes with feature call ##
14:20.05ziz212Once you in a call you can press ## and then you are asked to enter a extension. Once you enter call get transfer.
14:20.16SamotYou are going about this all wrong.
14:20.18[TK]D-Fenderthat is basic call transfer
14:20.23SamotYou are not understanding the concepts.
14:20.34ziz212I have tried to catch this and automate this
14:21.04SamotHuh?
14:21.17ziz212to a fixed canference
14:21.27[TK]D-Fenderwell if you wnt to LITERALLY use a normal "call transfer", then have your phone send the DTMF to use the Dial()-based featurecode and dial the exten you want to transfer to
14:21.42[TK]D-Fenderif you want AMI to do it then that is an APPLICATIONMAP and is something ENTIRELY different
14:22.08SamotPlus, FreePBX.
14:23.29[TK]D-FenderI'll wait until he's CLEAR about the exact methodology he's looking to implement before seeing if he is actually screwed by his environment
14:23.37[TK]D-FenderI jsut want a CLEAR description of the terms first
14:24.48[TK]D-Fender<ziz212> I need to blind transfer the call to meeting room once originator press dtmf sequence . <-- ## is DTMF for the DIAL based feature.  other digits AFTER that could be the TARGET of your transfer
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14:26.23ziz212How can we see what happen in back end of this blind transfer.? I have used asterisk -rvvvv but it didnt show the each instance how asterisk catch the dtmf
14:26.36ziz212##
14:27.00[TK]D-FenderYou LOOK
14:27.06[TK]D-FenderWhere are we looking at a call?
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14:28.36ziz212Asterisk shell
14:28.53[TK]D-Fenderwaits...
14:32.16ziz212Guys. I am really searching for this. If you can help me please guide.
14:32.23[TK]D-FenderWHERE'
14:32.29[TK]D-FenderWHERE'S THE CALL TO LOOK AT?
14:32.36[TK]D-FenderWhat part of "LOOK" are you having trouble with?
14:32.42ziz212asterisk -rvvvvvv
14:32.52ziz212Ok .
14:32.55ziz212I will tell
14:37.23ziz212In each incoming call or outgoing call, it follows in/out route and trunk OR otherway around. All are written in dialplan. Events one by one logged on asterisk log and visually can be seen in Asterisk CLI.
14:37.53[TK]D-Fender...
14:38.03ziz212All are written in dialplan >> need to be corrected as >> all follows dial plan
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14:38.59Samotgets [TK]D-Fender a REALLY BIG mug of coffee....
14:39.01[TK]D-FenderNone of that means anything
14:39.02|ance|ottHi guys I am running in to an issue with VM notifications on Asterisk 11, when Asterisk send the VM notification to emailA@mydomain.com,emailB@mydomain.com just first mail address got the attachment
14:39.15[TK]D-FenderAnd is not what was requested
14:39.23|ance|ottthis is something on the configuration that has to be change in order to all of them receive the attachment?
14:39.32|ance|ottany hint will be appreciate
14:39.35ziz212What I cannot see is >> blind transfer feature is not properly logged in asterisk CLI.
14:39.37[TK]D-Fender|ance|ott, You can't
14:39.58[TK]D-Fender|ance|ott, * will not send the full to multiple.  The config sample shos this.  one address only for full
14:40.20[TK]D-Fenderto send to multipel you'd have to target something that is counted as a distribution list by your MTA or the target
14:40.23|ance|ott@[TK]D-Fender  so I have to pacth Asterisk or so?
14:40.40[TK]D-Fendermake your target a group
14:40.49|ance|ottyeah i know that workaround
14:40.55|ance|ottbut we are trying to avoid that
14:41.25[TK]D-Fender|ance|ott, You've got that ... or you've got the source
14:41.33|ance|ottright :)
14:41.43|ance|ottthank you , I really appreciate your time
14:41.58[TK]D-Fender|ance|ott, or whatever consultant you can hire to do it....
14:42.26[TK]D-Fenderbut yeah, it's still not an option or something you can more easily work around by hack-ish means
14:43.59ziz212Is there any help for my work?
14:44.56ziz212Any support for me to do my work please
14:45.42[TK]D-Fenderziz212, WHERE'S THE FUCKING CALL I ASKED FOR 5 TIMES?
14:45.49Samot[TK]D-Fender: Multiple recipients are allowed for Voicemail()
14:46.21Samotemail1@domain.com|email2@domain.com works
14:46.36ziz212[TK]D-Fenderziz212, WHERE'S THE FUCKING CALL I ASKED FOR 5 TIMES? >>> I really dont know what you are asking from me.
14:46.45SamotAt least in 13 I've been doing it.
14:47.02ziz212I really would like to get your help
14:49.52[TK]D-FenderCLI output of your call <-------
14:49.55[TK]D-Fenderwhere is it?
14:50.05[TK]D-FenderSHOW US YOUR attempt TO do THIS THING YOU ARE ASKING ABOUT
14:50.41ziz212Ok pls let me share that.
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14:55.45[TK]D-Fender~pb
14:55.45infobotsomebody said pastebin was 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.
14:55.46[TK]D-Fender^^^
15:02.44ziz212http://paste.lisp.org/+7HEC/2.
15:13.08ziz212What I am trying to do is not just ## .. but ##9900 and call automatically get transfer to the room as blind transfer.
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15:15.17[TK]D-Fenderyou are doing a PHONE tansfer there
15:15.17[TK]D-Fender[2017-06-20 23:57:26] NOTICE[24515][C-00000001]: chan_sip.c:26219 handle_request_invite: Call from '100001' (112.134.2.84:65389) to extension '##9900' rejected because extension not found in context 'from-internal'.
15:15.29[TK]D-FenderYou do not TRANSFER to ##.
15:15.35[TK]D-Fender## is supposed to be DTMF <------
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15:15.46[TK]D-Fenderbut that's part of a SIP transfer and NOT a DTMF truiggered one
15:19.57ziz212Once I press ## on phone , a message play and ask me to enter number so I enter 9900 .
15:20.16ziz212then call get transferred. My phone get free
15:21.18[TK]D-Fender[2017-06-20 23:57:26] NOTICE[24515][C-00000001]: chan_sip.c:26219 handle_request_invite: Call from '100001' (112.134.2.84:65389) to extension '##9900' rejected because extension not found in context 'from-internal'.
15:21.27[TK]D-Fenderyou are transferring to ##9900
15:21.31[TK]D-Fenderyou told it ## in front
15:21.37[TK]D-Fenderyou should not have that
15:26.12ziz212it is a freepbx feature.. http://i.xomf.com/cxxls.jpg << screenshot
15:30.35[TK]D-Fenderyou are dialing ## IN the target
15:31.03ziz212Yes. First I dial ## then message plays. then I enter the 9900.
15:31.26ziz212then it try to see is it an extension.
15:31.42[TK]D-Fenderclearly oing it wrong
15:31.42ziz212once it found that it is not. it transfer the call to room 9900
15:31.52[TK]D-Fender<[TK]D-Fender> [2017-06-20 23:57:26] NOTICE[24515][C-00000001]: chan_sip.c:26219 handle_request_invite: Call from '100001' (112.134.2.84:65389) to extension '##9900' rejected because extension not found in context 'from-internal'.
15:32.38ziz212Yes, it is not found in context 'from-internal' because I never create that.
15:32.53[TK]D-FenderNO
15:33.02[TK]D-FenderCall from '100001' (112.134.2.84:65389) to extension '##9900' rejected because extension not found in context 'from-internal'.
15:33.09[TK]D-Fenderto extension '##9900'
15:33.54[TK]D-Fenderthere is ## in the TARGET
15:33.54ziz212But to come for blind transfer mode I need to press ##
15:33.56ziz212then it plays the message
15:34.01ziz212then I enter 9900
15:34.08[TK]D-FenderDoesn't look like that's the case
15:34.13ziz212that part is not logged in CLI
15:34.19[TK]D-FenderEnable core debug to prove the dtmf
15:34.28ziz212Sure
15:34.29[TK]D-Fendercore set debug 10
15:34.31[TK]D-Fendernew call
15:34.38ziz212OK
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15:47.46ziz212past is too large
15:47.52ziz212what can I do
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15:56.03ziz212@[TK]D-Fender : Past is too big. Pastbin does not allow to past it
16:04.03ziz212https://pastebin.com/0dzPtnGB                          https://pastebin.com/3crvW6Gx
16:04.34ziz212@[TK]D-Fender : Please be kind enough to have a look
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18:49.04tenspeed705looking for anyone who is using a cisco cp-78xx(7811 specifically) on asterisk
18:52.16salviadudtenspeed705, right here
18:52.22salviadudwait, no.
18:52.28salviadud79xx
18:52.30salviadudsorry
18:52.47salviadudIt might be similar, but I'm not entirely sure
18:53.18tenspeed705no, they are not...starting to wish it was a 79xx....:(
18:53.32salviadudAre you using sip or sccp?
18:54.06salviadudtenspeed705, so have you used a 79xx series on asterisk before, that's why you said that.
18:54.09tenspeed705trying to flash to SIP
18:54.28salviadudtenspeed705, what have you tried?
18:54.39tenspeed705yeah i got a mix of 794x 796x and 791x
18:55.44salviadudI imagine you have sipdefault.cnf inside your tftp boot dir
18:56.01tenspeed705sorry, the 7811 looks to already be using SIP but it has firmware for CUCM I am trying to flash to a MPP firmware (3ed party) to use in asterisk
18:56.23tenspeed705sipdefault.cnf is there
18:56.54salviadudThe image file inside sipdefault.cnf is it the thirdparty one?
18:57.07salviadudwell it's the image in text
18:58.37tenspeed705yeah its sip78xx.7-0-0MPP-7.loads
18:59.15salviadudtake out the .loads if its there
18:59.24salviadudit should be like: image_version: "SIP42.9-4-2SR3-1S"
19:00.14tenspeed705yeah, its like that
19:00.50salviadudWhat does the phone do?
19:00.57salviadudWhen you boot it.
19:02.35salviadudDoes it "complain"?
19:02.44tenspeed705GUI shows nothing at all. just REGISTERING. I do see it trying to download the firmware VIA FTP, then the phone changes to Last Upgrade(Failure)
19:03.42tenspeed705TFTP**
19:03.56salviadudI would try something like changing it back to original sccp and then flashing it to that third party sip
19:04.37tenspeed705tried. I downgraded as far as I could for CUCM. trying the MPP firmware between each downgrade.
19:04.48salviadudThen again, I have not dealt with the 78xx series, just 79 and up
19:04.58salviadudis the cisco sip firmware that bad? hehe
19:05.28tenspeed705I have never really liked it...seems a lot better on the SPA series tho
19:06.01salviadudWell, the cisco phones that only do sip, work pretty well yeah
19:06.16salviadudI got a 9971 myself
19:08.04tenspeed705for my acutall phone I use a GXV3275
19:10.03salviadudThat's a nice phone, do you use the camera too?
19:13.31tenspeed705i do at times. Not to many people around here with video phones tho
19:15.33salviadudI was trying to make video to work, but I read that I had to take out the use_callmanager option
19:15.59salviadudAnd I rather be able to do ad-hoc conferences than watch my boss, when he's just a few feet away.
19:16.02tenspeed705oh weird
19:22.04salviadudI wrongly assume that you have installed that patch from that page from new zealand
19:22.20salviadudI shouldn't assume
19:22.38salviadudBut you should look into it if you want full functionality for cisco via sip on asterisk.
19:22.52salviadudI don't think it works for pjsip though
19:23.16tenspeed705yeah, just using normal sip
19:24.46salviadudThe patch I mean: http://usecallmanager.nz/document-overview.html
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19:31.34tenspeed705nice. I will need to look in to that
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19:53.22salviadudtenspeed705, might make your 78xx work better with the native sip firmware
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22:04.51Nico67hello
22:06.44Nico67I'm coding for a taxi company, and we would like to detect with a nearly 100% success answering machines, talkdetect and AMD modules are not enough satisfying
22:07.35[TK]D-FenderThat's all we've got
22:07.56Nico67we already put a specific sound on taxi answer message (4 bips 440Hz), do you think it may be easily detected by asterisk ?
22:08.40[TK]D-Fender* doesn't detect anything
22:08.44[TK]D-FenderAPP do things
22:09.13[TK]D-FenderDo you see an app where you can plug in frequencies for a run-time determination?
22:09.20Nico67I tried to read dsp.c, app_amd.c, app_talker.c but a little bit complex ;)
22:09.35[TK]D-FenderSo far I don't see any
22:09.45[TK]D-FenderYou could write your own, and it's up to you if you wanted
22:10.51[TK]D-Fenderyou don't need to read AMD's source to see if it can do this.  It'd be in the INSTRUCTIONS
22:12.45Nico67where are INSTRUCTIONS ?
22:14.37[TK]D-Fender"core show application amd".  and in the config sample
22:14.37[TK]D-FenderThe options aren't exactly hidden
22:15.10Nico67ok, already read the ADM documentation many many times ;)

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