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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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| 01:31.23 | jeffspeff | https://www.youtube.com/watch?v=44uYz6PuTj0 |
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| 09:41.53 | ofg|brtr | hello people |
| 09:42.02 | [sID] | hi |
| 09:42.52 | ofg|brtr | I need to start a SIP server. I don't have any telephony hardware. What I have is a Virtual server with a valid IP. |
| 09:43.01 | ofg|brtr | How should I go about it. |
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| 09:43.27 | ofg|brtr | It would be very helpful if someone enlighten me. |
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| 09:55.00 | ofg|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 |
| 10:06.13 | ofg|brtr | [sID]: So I can setup a sip with freepbx. |
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| 11:48.24 | pawiecki | Anyone has some experience with KWS 6000? |
| 11:55.28 | pawiecki | hmm, nevermind, bad question. It looks like after importing config backup, KWS reverted to default settings, with red led flashing. |
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| 12:16.53 | pawiecki | probably interrupted backup file, because another one worked. Anyway not fun. |
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| 14:00.09 | ziz212 | Dear All, I am having problem writing bash AMI script with telnet |
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| 14:03.54 | pawiecki | ziz212: tell us WHAT is the problem - describe it. |
| 14:04.12 | Samot | ziz212: As [TK]D-Fender said in the other channel, AMI is not a language. |
| 14:04.39 | Samot | So you're writing a bash script that telnet's to Asterisk to issue an AMI command... |
| 14:04.56 | ziz212 | Yes exactly write. |
| 14:05.10 | Samot | And the problem is? |
| 14:05.26 | ziz212 | after the telnet session AMI commands are not running |
| 14:05.37 | ziz212 | script stop from there |
| 14:05.49 | Samot | Show the script |
| 14:05.50 | Samot | ~pb |
| 14:05.58 | infobot | somebody 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. |
| 14:05.58 | ziz212 | It wait for Stdin |
| 14:06.01 | Samot | Are you sending any? |
| 14:06.44 | ziz212 | Action: Login Username: testuser Secret: testsecret |
| 14:06.48 | Samot | OK |
| 14:06.51 | Samot | That logs you in. |
| 14:07.05 | ziz212 | It never execute after the telnet command |
| 14:07.10 | ziz212 | it waits for stdin |
| 14:07.16 | Samot | Right |
| 14:07.19 | ziz212 | not taking from the script next line |
| 14:07.25 | Samot | Show the script |
| 14:07.27 | Samot | ~pb |
| 14:07.28 | infobot | it 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. |
| 14:07.48 | pawiecki | I'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? |
| 14:08.32 | Samot | Never have I seen it where the MAC is the username. |
| 14:08.53 | Samot | In regards to the "sip account" for the device. |
| 14:09.04 | ziz212 | OK. http://paste.lisp.org/display/349140 |
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| 14:11.23 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:11.33 | [TK]D-Fender | <ziz212> It wait for Stdin <- no |
| 14:11.46 | [TK]D-Fender | AMI does NOT use stdin/sdtout |
| 14:12.00 | [TK]D-Fender | AMI is a TCP SOCKET CONNECTION |
| 14:13.20 | ziz212 | Once 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 |
| 14:13.53 | Samot | I don't use telnet with AMI but I'm guessing your cat should include all of it |
| 14:13.57 | Samot | In one |
| 14:13.59 | Samot | Not three. |
| 14:15.31 | Samot | But I could be wrong. |
| 14:15.40 | ziz212 | I am trying to add featuremap as shell script |
| 14:15.53 | Samot | OK. |
| 14:16.28 | ziz212 | I need to blind transfer the call to meeting room once originator press dtmf sequence . |
| 14:16.52 | Samot | Let's get past making the commands run first. |
| 14:17.03 | ziz212 | I edited the features.conf file and add the line. but I dont know how to do the blind transfer to that |
| 14:17.04 | Samot | Figure out how to hammer in the nail... |
| 14:17.06 | Samot | Then build the house. |
| 14:17.07 | ziz212 | it 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-Fender | You don't seem to understand any of the pieces you are using... |
| 14:18.05 | [TK]D-Fender | features EXECUTE on the channel they are bound to |
| 14:18.16 | [TK]D-Fender | there is no TRANSFER |
| 14:18.23 | [TK]D-Fender | You call whatever you have it call |
| 14:18.23 | ziz212 | I think I am wasting my time ... |
| 14:18.31 | [TK]D-Fender | I think you don't understand the words you're using |
| 14:18.32 | Samot | If you're originating a call to someone... |
| 14:18.37 | Samot | And they press 22... |
| 14:18.49 | Samot | You aren't transferring to them to conf bridge... |
| 14:18.57 | Samot | You are _sending_ them to it |
| 14:19.11 | Samot | You can put them right in it. |
| 14:19.37 | ziz212 | Please let me have some word for you. Freepbx comes with feature call ## |
| 14:20.05 | ziz212 | Once you in a call you can press ## and then you are asked to enter a extension. Once you enter call get transfer. |
| 14:20.16 | Samot | You are going about this all wrong. |
| 14:20.18 | [TK]D-Fender | that is basic call transfer |
| 14:20.23 | Samot | You are not understanding the concepts. |
| 14:20.34 | ziz212 | I have tried to catch this and automate this |
| 14:21.04 | Samot | Huh? |
| 14:21.17 | ziz212 | to a fixed canference |
| 14:21.27 | [TK]D-Fender | well 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-Fender | if you want AMI to do it then that is an APPLICATIONMAP and is something ENTIRELY different |
| 14:22.08 | Samot | Plus, FreePBX. |
| 14:23.29 | [TK]D-Fender | I'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-Fender | I 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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| 14:26.23 | ziz212 | How 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 |
| 14:26.36 | ziz212 | ## |
| 14:27.00 | [TK]D-Fender | You LOOK |
| 14:27.06 | [TK]D-Fender | Where are we looking at a call? |
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| 14:28.36 | ziz212 | Asterisk shell |
| 14:28.53 | [TK]D-Fender | waits... |
| 14:32.16 | ziz212 | Guys. I am really searching for this. If you can help me please guide. |
| 14:32.23 | [TK]D-Fender | WHERE' |
| 14:32.29 | [TK]D-Fender | WHERE'S THE CALL TO LOOK AT? |
| 14:32.36 | [TK]D-Fender | What part of "LOOK" are you having trouble with? |
| 14:32.42 | ziz212 | asterisk -rvvvvvv |
| 14:32.52 | ziz212 | Ok . |
| 14:32.55 | ziz212 | I will tell |
| 14:37.23 | ziz212 | In 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 | ... |
| 14:38.03 | ziz212 | All are written in dialplan >> need to be corrected as >> all follows dial plan |
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| 14:38.59 | Samot | gets [TK]D-Fender a REALLY BIG mug of coffee.... |
| 14:39.01 | [TK]D-Fender | None of that means anything |
| 14:39.02 | |ance|ott | Hi 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-Fender | And is not what was requested |
| 14:39.23 | |ance|ott | this is something on the configuration that has to be change in order to all of them receive the attachment? |
| 14:39.32 | |ance|ott | any hint will be appreciate |
| 14:39.35 | ziz212 | What 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-Fender | to 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-Fender | make your target a group |
| 14:40.49 | |ance|ott | yeah i know that workaround |
| 14:40.55 | |ance|ott | but 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|ott | right :) |
| 14:41.43 | |ance|ott | thank 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-Fender | but yeah, it's still not an option or something you can more easily work around by hack-ish means |
| 14:43.59 | ziz212 | Is there any help for my work? |
| 14:44.56 | ziz212 | Any support for me to do my work please |
| 14:45.42 | [TK]D-Fender | ziz212, WHERE'S THE FUCKING CALL I ASKED FOR 5 TIMES? |
| 14:45.49 | Samot | [TK]D-Fender: Multiple recipients are allowed for Voicemail() |
| 14:46.21 | Samot | email1@domain.com|email2@domain.com works |
| 14:46.36 | ziz212 | [TK]D-Fenderziz212, WHERE'S THE FUCKING CALL I ASKED FOR 5 TIMES? >>> I really dont know what you are asking from me. |
| 14:46.45 | Samot | At least in 13 I've been doing it. |
| 14:47.02 | ziz212 | I really would like to get your help |
| 14:49.52 | [TK]D-Fender | CLI output of your call <------- |
| 14:49.55 | [TK]D-Fender | where is it? |
| 14:50.05 | [TK]D-Fender | SHOW US YOUR attempt TO do THIS THING YOU ARE ASKING ABOUT |
| 14:50.41 | ziz212 | Ok pls let me share that. |
| 14:55.02 | *** join/#asterisk skywayskase (~skywayska@163.182.162.226) |
| 14:55.45 | [TK]D-Fender | ~pb |
| 14:55.45 | infobot | somebody 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 | ^^^ |
| 15:02.44 | ziz212 | http://paste.lisp.org/+7HEC/2. |
| 15:13.08 | ziz212 | What I am trying to do is not just ## .. but ##9900 and call automatically get transfer to the room as blind transfer. |
| 15:15.11 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 15:15.17 | [TK]D-Fender | you 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-Fender | You do not TRANSFER to ##. |
| 15:15.35 | [TK]D-Fender | ## is supposed to be DTMF <------ |
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| 15:15.38 | *** mode/#asterisk [+o wdoekes] by ChanServ |
| 15:15.46 | [TK]D-Fender | but that's part of a SIP transfer and NOT a DTMF truiggered one |
| 15:19.57 | ziz212 | Once I press ## on phone , a message play and ask me to enter number so I enter 9900 . |
| 15:20.16 | ziz212 | then 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-Fender | you are transferring to ##9900 |
| 15:21.31 | [TK]D-Fender | you told it ## in front |
| 15:21.37 | [TK]D-Fender | you should not have that |
| 15:26.12 | ziz212 | it is a freepbx feature.. http://i.xomf.com/cxxls.jpg << screenshot |
| 15:30.35 | [TK]D-Fender | you are dialing ## IN the target |
| 15:31.03 | ziz212 | Yes. First I dial ## then message plays. then I enter the 9900. |
| 15:31.26 | ziz212 | then it try to see is it an extension. |
| 15:31.42 | [TK]D-Fender | clearly oing it wrong |
| 15:31.42 | ziz212 | once 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'. |
| 15:32.38 | ziz212 | Yes, it is not found in context 'from-internal' because I never create that. |
| 15:32.53 | [TK]D-Fender | NO |
| 15:33.02 | [TK]D-Fender | Call from '100001' (112.134.2.84:65389) to extension '##9900' rejected because extension not found in context 'from-internal'. |
| 15:33.09 | [TK]D-Fender | to extension '##9900' |
| 15:33.54 | [TK]D-Fender | there is ## in the TARGET |
| 15:33.54 | ziz212 | But to come for blind transfer mode I need to press ## |
| 15:33.56 | ziz212 | then it plays the message |
| 15:34.01 | ziz212 | then I enter 9900 |
| 15:34.08 | [TK]D-Fender | Doesn't look like that's the case |
| 15:34.13 | ziz212 | that part is not logged in CLI |
| 15:34.19 | [TK]D-Fender | Enable core debug to prove the dtmf |
| 15:34.28 | ziz212 | Sure |
| 15:34.29 | [TK]D-Fender | core set debug 10 |
| 15:34.31 | [TK]D-Fender | new call |
| 15:34.38 | ziz212 | OK |
| 15:45.39 | *** join/#asterisk danjenkins (danjenkins@gateway/shell/firrre/x-sdsrjjqeiemdqvxo) |
| 15:45.39 | *** mode/#asterisk [+o danjenkins] by ChanServ |
| 15:47.46 | ziz212 | past is too large |
| 15:47.52 | ziz212 | what can I do |
| 15:48.33 | *** join/#asterisk salviadud (~ralfalfa@189-211-190-134.static.axtel.net) |
| 15:56.03 | ziz212 | @[TK]D-Fender : Past is too big. Pastbin does not allow to past it |
| 16:04.03 | ziz212 | https://pastebin.com/0dzPtnGB https://pastebin.com/3crvW6Gx |
| 16:04.34 | ziz212 | @[TK]D-Fender : Please be kind enough to have a look |
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| 18:48.31 | *** join/#asterisk tenspeed705 (~keiths@sud-fw01-inet-wan.vianet.ca) |
| 18:49.04 | tenspeed705 | looking for anyone who is using a cisco cp-78xx(7811 specifically) on asterisk |
| 18:52.16 | salviadud | tenspeed705, right here |
| 18:52.22 | salviadud | wait, no. |
| 18:52.28 | salviadud | 79xx |
| 18:52.30 | salviadud | sorry |
| 18:52.47 | salviadud | It might be similar, but I'm not entirely sure |
| 18:53.18 | tenspeed705 | no, they are not...starting to wish it was a 79xx....:( |
| 18:53.32 | salviadud | Are you using sip or sccp? |
| 18:54.06 | salviadud | tenspeed705, so have you used a 79xx series on asterisk before, that's why you said that. |
| 18:54.09 | tenspeed705 | trying to flash to SIP |
| 18:54.28 | salviadud | tenspeed705, what have you tried? |
| 18:54.39 | tenspeed705 | yeah i got a mix of 794x 796x and 791x |
| 18:55.44 | salviadud | I imagine you have sipdefault.cnf inside your tftp boot dir |
| 18:56.01 | tenspeed705 | sorry, 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 |
| 18:56.23 | tenspeed705 | sipdefault.cnf is there |
| 18:56.54 | salviadud | The image file inside sipdefault.cnf is it the thirdparty one? |
| 18:57.07 | salviadud | well it's the image in text |
| 18:58.37 | tenspeed705 | yeah its sip78xx.7-0-0MPP-7.loads |
| 18:59.15 | salviadud | take out the .loads if its there |
| 18:59.24 | salviadud | it should be like: image_version: "SIP42.9-4-2SR3-1S" |
| 19:00.14 | tenspeed705 | yeah, its like that |
| 19:00.50 | salviadud | What does the phone do? |
| 19:00.57 | salviadud | When you boot it. |
| 19:02.35 | salviadud | Does it "complain"? |
| 19:02.44 | tenspeed705 | GUI 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) |
| 19:03.42 | tenspeed705 | TFTP** |
| 19:03.56 | salviadud | I would try something like changing it back to original sccp and then flashing it to that third party sip |
| 19:04.37 | tenspeed705 | tried. I downgraded as far as I could for CUCM. trying the MPP firmware between each downgrade. |
| 19:04.48 | salviadud | Then again, I have not dealt with the 78xx series, just 79 and up |
| 19:04.58 | salviadud | is the cisco sip firmware that bad? hehe |
| 19:05.28 | tenspeed705 | I have never really liked it...seems a lot better on the SPA series tho |
| 19:06.01 | salviadud | Well, the cisco phones that only do sip, work pretty well yeah |
| 19:06.16 | salviadud | I got a 9971 myself |
| 19:08.04 | tenspeed705 | for my acutall phone I use a GXV3275 |
| 19:10.03 | salviadud | That's a nice phone, do you use the camera too? |
| 19:13.31 | tenspeed705 | i do at times. Not to many people around here with video phones tho |
| 19:15.33 | salviadud | I was trying to make video to work, but I read that I had to take out the use_callmanager option |
| 19:15.59 | salviadud | And I rather be able to do ad-hoc conferences than watch my boss, when he's just a few feet away. |
| 19:16.02 | tenspeed705 | oh weird |
| 19:22.04 | salviadud | I wrongly assume that you have installed that patch from that page from new zealand |
| 19:22.20 | salviadud | I shouldn't assume |
| 19:22.38 | salviadud | But you should look into it if you want full functionality for cisco via sip on asterisk. |
| 19:22.52 | salviadud | I don't think it works for pjsip though |
| 19:23.16 | tenspeed705 | yeah, just using normal sip |
| 19:24.46 | salviadud | The patch I mean: http://usecallmanager.nz/document-overview.html |
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| 19:31.34 | tenspeed705 | nice. I will need to look in to that |
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| 19:53.22 | salviadud | tenspeed705, might make your 78xx work better with the native sip firmware |
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| 22:04.48 | *** join/#asterisk Nico67 (4ec66b44@gateway/web/freenode/ip.78.198.107.68) |
| 22:04.51 | Nico67 | hello |
| 22:06.44 | Nico67 | I'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-Fender | That's all we've got |
| 22:07.56 | Nico67 | we 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-Fender | APP do things |
| 22:09.13 | [TK]D-Fender | Do you see an app where you can plug in frequencies for a run-time determination? |
| 22:09.20 | Nico67 | I tried to read dsp.c, app_amd.c, app_talker.c but a little bit complex ;) |
| 22:09.35 | [TK]D-Fender | So far I don't see any |
| 22:09.45 | [TK]D-Fender | You could write your own, and it's up to you if you wanted |
| 22:10.51 | [TK]D-Fender | you don't need to read AMD's source to see if it can do this. It'd be in the INSTRUCTIONS |
| 22:12.45 | Nico67 | where are INSTRUCTIONS ? |
| 22:14.37 | [TK]D-Fender | "core show application amd". and in the config sample |
| 22:14.37 | [TK]D-Fender | The options aren't exactly hidden |
| 22:15.10 | Nico67 | ok, already read the ADM documentation many many times ;) |
