| 00:00.43 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 00:05.55 | *** join/#asterisk jetlag (~jetlag@c-71-226-222-56.hsd1.nj.comcast.net) |
| 00:23.17 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:23.17 | *** 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 |
| 00:38.53 | *** join/#asterisk phix (~threat@210.185.101.58) |
| 00:57.03 | *** join/#asterisk jetlag (~jetlag@c-71-226-222-56.hsd1.nj.comcast.net) |
| 01:06.47 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 01:23.27 | *** join/#asterisk wh00 (43f6e942@gateway/web/cgi-irc/kiwiirc.com/ip.67.246.233.66) |
| 01:24.00 | wh00 | Hi All, is there a way to capture ${DIALSTATUS} when using .call files ? |
| 01:24.19 | Samot | The same way you capture with any other Dial() |
| 01:25.18 | wh00 | Always returns an empty string |
| 01:25.29 | Samot | Show it... |
| 01:25.39 | Samot | .call files just originate a call |
| 01:25.44 | Samot | Sends them to a context.. |
| 01:25.54 | Samot | Dial doesn't have to be part of that |
| 01:25.57 | Samot | Honestly.. |
| 01:26.15 | Samot | asterisk -rvvvvvvvvvvvv |
| 01:26.24 | Samot | generate call via .call file and pastebin the results. |
| 01:26.27 | Samot | ~pb |
| 01:26.27 | infobot | pastebin is probably 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. |
| 01:34.24 | wh00 | https://pastebin.com/UWSSDGZH |
| 01:36.07 | Samot | Where's the Dial() |
| 01:36.18 | Samot | DIALSTATUS is a result of a Dial(). |
| 01:36.22 | Samot | You just can't call on it. |
| 01:37.44 | Samot | So it doesn't return until the Dial() is completed, i.e. the call itself. |
| 01:37.54 | Samot | You can't get the DIALSTATUS will on the call. |
| 01:40.02 | Samot | s/will/while/ |
| 01:40.02 | wh00 | How does the call get made when using a .call file? When I move the .call file to outgoing it makes the call without any dial call. |
| 01:40.21 | Samot | Call files are nothing more than an Originate. |
| 01:40.50 | Samot | You send the call to TECH/exten@context |
| 01:41.17 | Samot | The Context settings is the context to execute when the Dial() is answered. |
| 01:41.27 | Samot | Generally it's Local/exten@context |
| 01:42.01 | Samot | Because when a Local channel issues a Dial() that is answered it switches the direction of the call |
| 01:42.16 | Samot | There are two legs to a Local channel, outbound and inbound. |
| 01:42.58 | Samot | And since you're using FreePBX there's a commercial module for this already |
| 01:43.06 | Samot | "Broadcast" |
| 01:43.18 | Samot | Does what you are looking to do plus more. |
| 01:44.16 | wh00 | So it sounds like my call file is setup incorrectly then? |
| 01:44.42 | Samot | Or the context that handles it |
| 01:44.43 | Samot | Or both |
| 01:44.44 | *** join/#asterisk spicyramen (~Adium@c-73-241-232-155.hsd1.ca.comcast.net) |
| 01:45.05 | Samot | Again, you should just get the licence for Broadcast |
| 01:45.19 | Samot | It does exactly what you are looking to do. |
| 01:46.45 | Samot | I'd go look at that and see if it meets your needs at an acceptable level. |
| 01:47.11 | wh00 | We did look at it. Didn't seem to be what we were looking for. |
| 01:47.33 | Samot | You want people to interact? |
| 01:48.17 | wh00 | That is one of the things. |
| 01:48.41 | Samot | If you're using AMD and expect a DTMF or some response from the callee, you need to start with a Local channel that will use a SIP channel to dialout the SIP trunk |
| 01:49.03 | wh00 | The whole app already works, just doesn't have the dial status at the end. |
| 01:49.18 | Samot | You're missing the point |
| 01:49.40 | wh00 | Correct. Please guide me. |
| 01:50.32 | Samot | Channel: Local/1NXXNXXXXXX@outbound/n <-- That |
| 01:50.52 | Samot | Channel: SIP/1NXXNXXXXXX@siptrunk/n <-- If it's that, it's wrong. |
| 01:51.24 | Samot | TECH/EXTEN@CONTEXT |
| 01:51.28 | Samot | That's what it needs to be. |
| 01:51.44 | Samot | Not TECH/EXTEN@TRUNK/IP |
| 01:51.55 | wh00 | Ahhhh I see. I have SIP/channel/1NXXNXXXXXX I'll fix it and try again |
| 01:52.07 | Samot | That's not a "Dial" |
| 01:52.47 | Samot | Channel: Local/1NXXNXXXXXX@outbound/n <-- This opens a channel and sends the call to the EXTEN,Context,1 |
| 01:53.04 | Samot | Extension: s |
| 01:53.04 | Samot | Context: detect |
| 01:53.04 | Samot | Priority: 1 |
| 01:53.12 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 01:53.34 | Samot | ^^ Those are where to send the call if a Dial() is issues and answered from the original "Channel" location. |
| 01:53.52 | wh00 | Ok so now in the plan I add in the dial correct? |
| 01:54.06 | Samot | Your DIALSTATUS has to be AFTER the Dial() in the "outbound context" |
| 01:56.35 | Samot | TECH/EXTEN@CONTEXT <-- Think of this as the "endpoint" making the call. |
| 01:56.43 | Samot | It's just Asterisk doing it. |
| 01:56.52 | Samot | Instead of a SIP phone. |
| 01:57.53 | Samot | As you can tell by the readiness of that call file example, I do this already. |
| 02:02.44 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 02:03.03 | wh00 | Is 'outbound' the trunk? |
| 02:10.59 | wh00 | I must have something wrong. I keep getting "No such extension/context 1NXXNXXXXXX@outbound while calling Local channel' |
| 02:11.07 | *** join/#asterisk lankanmon_ (~LKNnet@2607:fea8:d1f:ffcb:4ce0:6e12:e323:6b6e) |
| 02:20.20 | Samot | Show the callfile |
| 02:24.11 | wh00 | https://thepasteb.in/p/3lh7N5vyoWli1 |
| 02:24.44 | Samot | That's not even complete. |
| 02:25.14 | Samot | You're missing the Priority: setting |
| 02:26.07 | wh00 | Sorry yes that is in there, just didn't copy/paste it is the last line. |
| 02:26.11 | Samot | Show the outbound context.. |
| 02:28.29 | wh00 | https://pastebin.com/caH6H1KW |
| 02:30.19 | Samot | Is that the outbound context? |
| 02:30.31 | Samot | And stop writing "etc" |
| 02:30.33 | Samot | Show it ALL |
| 02:30.43 | Samot | The outbound context is the context called outbound |
| 02:30.46 | Samot | [outbound] |
| 02:31.07 | Samot | Channel: Local/[Phone_number]@outbound/n <<--- EXTEN@CONTEXT |
| 02:31.17 | Samot | You have no outbound context. |
| 02:31.19 | wh00 | I don't believe there is an outbound. I just grepped the /etc/asterisk folder |
| 02:31.23 | Samot | No. |
| 02:31.26 | Samot | You have to WRITE it |
| 02:31.37 | Samot | It's the CONTEXT the call starts from. |
| 02:31.41 | wh00 | lol. There isn't one built in? |
| 02:31.45 | Samot | No. |
| 02:31.51 | Samot | Not at all |
| 02:31.58 | Samot | What I showed you is what *I* use |
| 02:32.06 | Samot | I have a context called [outbound] |
| 02:32.13 | wh00 | ok this is making more sense. |
| 02:32.40 | wh00 | So the next question is should I use my campaign context as the outbound, or should they be two totally separate contexts? |
| 02:32.46 | Samot | Did you not read up on callfiles and originate? |
| 02:32.50 | Samot | This is all explained. |
| 02:32.56 | wh00 | I did for days. |
| 02:32.58 | Samot | You need TWO |
| 02:33.13 | Samot | I don't know how many ways I can explain this.. |
| 02:33.31 | Samot | Channel: Local/[Phone_number]@outbound/n <--Actual context to start the call |
| 02:33.36 | Samot | Where the Dial() should live |
| 02:33.45 | Samot | And where DIALSTATUS will be issud. |
| 02:34.03 | Samot | Context: campaign-caller-1 |
| 02:34.03 | Samot | Extension: s |
| 02:34.03 | Samot | Priority: 1 |
| 02:34.13 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 02:34.21 | Samot | ^^ That's where to send an ANSWERED call from the Dial() |
| 02:34.35 | Samot | One context to make the Dial() |
| 02:34.43 | Samot | Another context to handle the call |
| 02:34.51 | Samot | Local channels are TWO leg channels. |
| 02:35.05 | Samot | Outbound (to make the call) and Inbound (answered call) |
| 02:35.30 | Samot | Once the call is answered it treats it like the callee was the one that MADE the call not ANSWERED the call. |
| 02:36.39 | wh00 | Ok I'm almost there. (absorbing all the information) |
| 02:37.13 | Samot | So [outbound] or whatever you want to call is where the call _starts_ |
| 02:37.27 | *** join/#asterisk lankanmon (~LKNnet@2607:fea8:d1f:ffcb:a51f:72b4:11ad:30be) |
| 02:37.52 | Samot | So it should have the exten => _1NXXNXXXXXX,n,Dial(stuff) |
| 02:38.05 | Samot | towards the end or basically the only thing there.. |
| 02:38.25 | Samot | Followed by any dialplan to handle the ${DIALSTATUS} variable returned |
| 02:47.37 | wh00 | Sooo you're going to love this. Everything appears to be working... but the dial status is still blank. |
| 02:57.55 | Samot | Then something is still wrong |
| 02:57.58 | Samot | I do this all the time. |
| 02:58.05 | Samot | I always get a DIALSTATUS |
| 02:58.12 | wh00 | I'm sure it's not you |
| 02:58.13 | wh00 | lol |
| 02:58.52 | wh00 | my NoOp also isn't printing in the log. |
| 02:59.10 | Samot | So your updated dialplan |
| 02:59.16 | Samot | Show |
| 03:00.18 | wh00 | [outbound] |
| 03:00.18 | wh00 | exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@[Trunk]) |
| 03:00.18 | wh00 | exten => _1NXXNXXXXXX,n,NoOp(Dialstatus is: ${DIALSTATUS}) ; Originally used same => but that didn't work either |
| 03:00.18 | wh00 | [outbound] |
| 03:00.18 | wh00 | exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@[Trunk]) |
| 03:00.18 | wh00 | exten => _1NXXNXXXXXX,n,NoOp(Dialstatus is: ${DIALSTATUS}) ; Originally used same => but that didn't work either |
| 03:00.57 | Samot | Show the call debug for a call |
| 03:04.45 | wh00 | https://pastebin.com/65yaahyJ |
| 03:06.53 | Samot | 10:30:30 PM <Samot> And stop writing "etc" |
| 03:06.53 | Samot | 10:30:32 PM <Samot> Show it ALL |
| 03:09.26 | Samot | Are you kidding? |
| 03:13.41 | *** join/#asterisk wh00 (43f6e942@gateway/web/cgi-irc/kiwiirc.com/ip.67.246.233.66) |
| 03:13.45 | wh00 | https://pastebin.com/Kj5JhEiH |
| 03:14.21 | wh00 | Does there need to be an Answer()? |
| 03:16.34 | *** join/#asterisk jetlag (~jetlag@c-71-226-222-56.hsd1.nj.comcast.net) |
| 03:20.39 | Samot | exten => h,n,NoOp(Dialstatus is: ${ORIGINATE_STATUS}) <-- That variable is never set |
| 03:23.00 | Samot | Also, you're handling the hangup within the call dialplan, it will never continue on after the Dial() in [outbound] context |
| 03:23.36 | Samot | wh00: It's very simple if the call hits [campaign-caller-1] the DIALSTATUS = 16 (Answered) |
| 03:23.52 | Samot | There's no need to look for the DIALSTATUS or guess the DIALSTATUS. |
| 03:24.04 | Samot | Because you're in a call, therefore it's Answered. |
| 03:24.39 | wh00 | yea all of that I'm already tracking. The only thing that I don't get is I get a timeout on busy signals instead of a busy response. |
| 03:24.54 | wh00 | I was hoping the dialstatus would contain if it was a busy signal. |
| 03:25.17 | Samot | Then you need to call a destination you know will be bust. |
| 03:25.20 | Samot | Then you need to call a destination you know will be busy. |
| 03:25.25 | Samot | Or won't answer |
| 03:28.40 | wh00 | Yea, when I do that I still get an empty dialstatus, hangup 0 and reason 3 |
| 03:31.40 | Samot | The DIALSTATUS is numeric |
| 03:31.53 | Samot | Oh wait.. |
| 03:31.54 | Samot | No. |
| 03:32.01 | Samot | I'm thinking of HANGUPCAUSE |
| 03:32.22 | Samot | That's actually probably want you want more than DIALSTATUS |
| 03:33.24 | wh00 | I have that already. Perhaps it is sufficient. |
| 03:33.34 | Samot | well you have to map it |
| 03:33.59 | Samot | And DIALSTATUS is a channel variable so you have to grab it from the channel that Dialed() |
| 03:34.21 | Samot | So you'll probably need a macro in your Dial() command to get it. |
| 03:35.08 | Samot | CONGESTION as a DIALSTATUS answer doesn't tell you anything. |
| 03:35.29 | Samot | It's used for various ISDN/Hangup cause codes |
| 03:35.40 | Samot | Which in turn are mapped from multiple SIP codes. |
| 03:35.41 | wh00 | I thought there was a busy status though? |
| 03:35.45 | Samot | There is |
| 03:36.00 | Samot | Which is the result of three or four different HANGUPCAUSE codes |
| 03:36.13 | wh00 | ooooh |
| 03:36.20 | Samot | <PROTECTED> |
| 03:36.22 | wh00 | I bet one of them is the timeout code that I'm already getting. |
| 03:36.31 | Samot | 408 is a timeout |
| 03:38.38 | wh00 | Hmm I'm getting a 0 for the hangup cause, but a 3 for the fail code. |
| 03:38.39 | Samot | But are they actual timeouts as in the endpoint didn't respond or are they timeouts because the endpoint just didn't answer... |
| 03:39.43 | Samot | 0 is unspecified error |
| 03:39.49 | Samot | 3 is no route to destination |
| 03:40.05 | Samot | What failed code? |
| 03:40.13 | Samot | Where are you getting that? |
| 03:40.33 | wh00 | ${REASON} |
| 03:42.14 | Samot | OK that's a dial timeout |
| 03:42.25 | Samot | The otherside isn't responding |
| 03:43.13 | wh00 | makes sense. I'm heading off for the night. Thank you so much for the help! |
| 03:43.23 | lorsungcu | np |
| 04:17.42 | *** join/#asterisk spicyramen (~Adium@c-73-241-232-155.hsd1.ca.comcast.net) |
| 04:18.17 | *** join/#asterisk spicyramen (~Adium@c-73-241-232-155.hsd1.ca.comcast.net) |
| 05:00.36 | *** join/#asterisk jetlag (~jetlag@c-71-226-222-56.hsd1.nj.comcast.net) |
| 05:49.53 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 06:02.50 | *** join/#asterisk jetlag (~jetlag@c-71-226-222-56.hsd1.nj.comcast.net) |
| 06:11.22 | *** join/#asterisk sip (~seattle@debug.siptool.com) |
| 06:24.30 | *** join/#asterisk sip (~seattle@debug.siptool.com) |
| 07:25.46 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:33.00 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 07:40.00 | *** join/#asterisk Mr_Pleb_Mgoo (~jakeb@103.46.213.148) |
| 08:47.53 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:00.02 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 09:05.15 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:52.08 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
| 10:03.29 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 10:30.35 | *** join/#asterisk ziz212 (~ziz212@175.157.183.1) |
| 10:32.22 | ziz212 | Hi, I need to find out how Blind transfer feature works in asterisk. I have increase the verbosity and debug levels but cannot identify how it is working. My target is to create same function Please help me in finding that |
| 10:32.29 | *** join/#asterisk miralin (~Thunderbi@85.115.248.14) |
| 10:40.22 | *** join/#asterisk miralin (~Thunderbi@95.153.135.141) |
| 10:46.19 | *** join/#asterisk miralin1 (~Thunderbi@85.115.248.14) |
| 11:40.22 | *** join/#asterisk marlinc_ (~marlinc@1.0.0.127.13.204.167.185.in-addr.arpa) |
| 12:44.57 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 12:48.33 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 12:59.20 | *** join/#asterisk Martin` (~martin@shell.ipv6.octocore.net) |
| 13:48.55 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 14:57.14 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 17:09.49 | *** join/#asterisk elegast (~elegast@5.79.86.129) |
| 17:32.23 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 17:46.24 | *** join/#asterisk aness (~aness@cm-84.209.155.137.getinternet.no) |
| 17:51.52 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
| 18:14.32 | *** join/#asterisk miralin (~Thunderbi@91.237.94.9) |
| 18:15.25 | *** join/#asterisk miralin (~Thunderbi@91.237.94.9) |
| 19:14.24 | igcewieling | I love The Onion. "Quantum Slacks, pants which are both formal and informal at the same time." |
| 19:41.10 | *** join/#asterisk spicyramen (~Adium@c-73-241-232-155.hsd1.ca.comcast.net) |
| 20:31.33 | *** join/#asterisk matrix1233 (~matrix123@41.226.243.129) |
| 20:44.27 | *** join/#asterisk kchehab (~denis@94.187.119.103) |
| 20:45.11 | kchehab | hi, how can enable logging the cancel calls in asteriskcdrdb |
| 20:49.38 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 20:57.08 | *** join/#asterisk jkroon (~jkroon@165.16.204.174) |
| 21:14.10 | kchehab | is there any place i can config to enbale stroing cancel calls |
| 21:15.09 | kchehab | or i should check call status in dialplan and insert it ? |
| 21:29.17 | *** join/#asterisk themayor (~themayor@unaffiliated/themayor) |
| 21:54.45 | *** join/#asterisk gusto (~gusto@2a01:c844:1046:820:eaee:4db2:8eb6:315d) |
| 22:02.49 | *** join/#asterisk elegast (~elegast@5.79.86.129) |
| 22:07.02 | *** join/#asterisk miralin (~Thunderbi@91.237.94.9) |
| 22:28.24 | *** join/#asterisk Iamnacho (~Iamnacho@ip70-171-163-5.om.om.cox.net) |
| 23:02.05 | *** join/#asterisk lvlinux (~ruel@unaffiliated/lvlinux) |
| 23:44.07 | *** join/#asterisk cool_boy (~chatzilla@179.183.175.95) |
| 23:45.12 | cool_boy | hi there, I am trying to make a call using a third party software(written in python) but getting ActionFailed: py-Asterisk action failed: Extension does not exist. |
| 23:45.34 | cool_boy | do I have to install py-Asterisk extension? where can I find that? |
| 23:45.58 | cool_boy | I tried to search but didn't find more than a python library pyastrisk, that is already install |
| 23:46.01 | cool_boy | installed * |
| 23:46.17 | Samot | Show the actual debug of the call and it failing. |
| 23:49.36 | cool_boy | Samot: https://pastebin.com/1drKeAgE |
| 23:50.06 | cool_boy | those are server logs of 3rd party software, I don't see any log on asterisk because request is not sent |
| 23:50.12 | Samot | I mean the Asterisk log |
| 23:50.12 | cool_boy | server breaks even before |
| 23:50.13 | Samot | For the call |
| 23:50.19 | Samot | Oh well then.. |
| 23:50.29 | Samot | This is third party software, you need to ask their support. |
| 23:50.40 | Samot | If the call isn't making it to Asterisk, then can't really help. |
| 23:50.48 | cool_boy | Samot: okay thanks |
| 23:51.05 | cool_boy | Samot: is there list of extensions that we can use in asterisk? |
| 23:52.05 | Samot | "Extensions" in this case is the actual user/device on Asterisk. |
| 23:52.13 | cool_boy | although I am not sure, here extension means a number assigned to agent or a missing library |
| 23:52.16 | Samot | Does the extension actually exist on Asterisk? Did you create it? |
| 23:52.24 | cool_boy | ohh thanks |
| 23:52.32 | Samot | Really? |
| 23:52.33 | cool_boy | yes it exists |
| 23:52.36 | Samot | OK |
| 23:52.39 | cool_boy | I created |
| 23:52.50 | Samot | Well then you need to get on the Odoo forums. |
| 23:53.01 | cool_boy | thanks a lot |
| 23:53.16 | Samot | py-Asterisk is their thing. |