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| 00:18.50 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); 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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| 01:07.01 | dl12 | Anyone run asterisk on hyperv? |
| 01:12.50 | Samot | Im sure there are. |
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| 02:20.58 | dl12 | Asterisk on hyperV ? |
| 02:21.07 | dl12 | Is it sta |
| 02:31.20 | [TK]D-Fender | Yes people run it on HyperV. |
| 02:31.22 | [TK]D-Fender | It can work |
| 02:31.27 | [TK]D-Fender | as well or as bad as anything else |
| 02:31.31 | [TK]D-Fender | it isn't magic. |
| 02:31.33 | [TK]D-Fender | NEXT!@!!! |
| 02:31.48 | dl12 | Haha |
| 02:32.09 | dl12 | I may switch from vmware to hyperv |
| 02:32.21 | dl12 | Just want to make sure! I will run some test first |
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| 07:26.54 | iBurger21 | Hey Samot; you were totally right about the NAT / or phone. I messed around in the settings (both firewarll and phone) and my Aastra is connectable now! |
| 07:27.40 | iBurger21 | I should go the extra mile to figure what really was the issue, (NAT or phone), but I've been a bit lazy; and just enjoyed the fact that I have a phone now that works. Which is tottaly f-ing awesome. :D |
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| 12:05.01 | jcpeters01 | I have a hard coded PAI to identify pilot number on a SIP trunk. That is currently what is seen on the Caller ID - is it possible to send the Pilot number PAI and then follow that up with the Remote Party ID? |
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| 12:20.14 | tzafrir | I didn't know Goto accepted (and ignored) "subroutine" arguments just like Gosub |
| 12:21.01 | tzafrir | That is: Goto(a_context,an_exten,1(whatever , random junk) is valid |
| 12:21.07 | tzafrir | err: |
| 12:21.17 | tzafrir | That is: Goto(a_context,an_exten,1(whatever , random junk)) is valid |
| 12:21.24 | tzafrir | Our use case: |
| 12:21.49 | tzafrir | Goto(a_context,an_exten,1(Set(whatever)) |
| 12:22.43 | tzafrir | err: that is: SET (the function) |
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| 13:38.05 | tafa2 | has anyone ever used https://www.didww.com/ ? |
| 13:42.57 | [TK]D-Fender | yes, people have |
| 13:43.00 | [TK]D-Fender | ~polls |
| 13:43.01 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 13:47.34 | tafa2 | grah |
| 13:47.54 | tafa2 | im trying to find out if they're a good service and if anyone in the channel would personally recommend them |
| 13:48.28 | [TK]D-Fender | The just ask those questions directly. |
| 13:48.42 | [TK]D-Fender | Nobody tends to come in saying "I love those guys" |
| 13:48.50 | [TK]D-Fender | (didww specifically) |
| 13:49.22 | [TK]D-Fender | most bringing up who they're using at all is just in reference to "I'm trying to configure PROVIDER and I'm screwing up, help!" |
| 13:49.29 | [TK]D-Fender | They work. |
| 13:50.03 | [TK]D-Fender | Nothing great or special from anything I've ever heard anyone say about the, |
| 13:51.15 | tafa2 | The just ask those questions directly. <--- noted |
| 13:52.02 | tafa2 | I'm basically trying to find DID's in the Middle East - UAE or Qatar |
| 13:52.29 | [TK]D-Fender | Asking leading questions can lead to repeat Q&A that gets you nowhere near where you actualy want to go. |
| 13:52.42 | [TK]D-Fender | So cut to the important part |
| 13:53.01 | [TK]D-Fender | <PROTECTED> |
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| 14:26.20 | muAdmDev | First time I got users from outside using my asterisk. Their IP is masqueraded (NAT). Any chance getting this to work with asterisk settings like 'externalip', without setting up a STUN server? |
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| 14:29.29 | [TK]D-Fender | yes, when you use the parameter names |
| 14:29.50 | [TK]D-Fender | right* |
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| 14:32.03 | martinvw | Hi, do I assume correctly that the System() app is synchronous and that Asterisk will wait for my external command to complete? Is there a similar app that won't wait for my external command? |
| 14:32.50 | igcewieling | martinvw: yes, but if the program becomes a daemon or otherwise goes into the background then the dialplan could continue. |
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| 14:33.00 | [TK]D-Fender | pretty much all dialplan apps except Background() are synchronous |
| 14:33.16 | [TK]D-Fender | So yes it can hold things up depending what you're doing. |
| 14:33.18 | martinvw | igcewieling: ok so the correct solution would be to write my external program so that it'll fork, so that Asterisk can continue as fast as possible? |
| 14:33.57 | igcewieling | martinvw: that is how I do it with my AGI scripts and I don't see any reason System() would be any different. |
| 14:34.10 | martinvw | OK, thanks for the advice! :) |
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| 15:48.39 | maurice2k | hi there, i have a question regarding ARI applications: is this still "the way to go"? i see a lot of outdated code |
| 15:49.24 | maurice2k | especially "asterisk/node-ari-client" is using some old swagger libs |
| 15:49.40 | maurice2k | triggering warnings |
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| 17:35.24 | martinvw | hm, I don't get how quoting variables is supposed to work in the dialplan |
| 17:35.51 | martinvw | I have a Dial call that looks like this: same => n,Dial(${SUPPORTPHONES},25,U(crmaddcontact^${CALLERID(num)}^${CALLERID(name)})) |
| 17:36.13 | martinvw | and in crmaddcontact, I try to print the passed values like this: same => n,NoOp(ARG2 = ${ARG2}) |
| 17:36.13 | file | maurice2k: depends on what you are trying to do, it is certainly a way to build telephony applications and is being used by people |
| 17:36.59 | martinvw | but in the output, Asterisk will cut CALLERID(name) off at the comma; so if the name was "Lastname, Firstname", NoOp() will only print "ARG2 = Lastname" |
| 17:37.15 | martinvw | I suppose I'd have to quote it somehow, but putting it in "" doesn't seem to do the trick :/ |
| 17:38.15 | martinvw | I'm not even sure if my mistake is in the Dial call (do I need to quote ${CALLERID(name)} before passing it as an argument?), or in my crmaddcontact sub |
| 17:45.40 | martinvw | ${LEN(${ARG2})} returns only the length of Lastname, which leads me to believe that the mistake must be in the Dial() call |
| 17:49.18 | igcewieling | martinvw: When I have an issue with commas in the data, I convert them to ! before passing the data to a macro or gosub, then convert it back to commas inside the gosub. |
| 17:50.20 | igcewieling | this is ael, but close enough. Set(dest=${REPLACE(dest,!,,)}); goto ${dest}; |
| 17:51.28 | igcewieling | I'd put a Dumpchan() in your crmaddcontact subroutine to see how the data is REALLY received. |
| 17:52.43 | martinvw | igcewieling: hm, that seems kind of icky to me. I was hoping that there might be a proper way to quote/escape variables so that I can pass them around easily. I mean, what if not's only commas that are interpreted differently than I expect, what if it'll unexpectedly interpret things like ${SHELL(evil injection attack)} too? :/ |
| 17:53.16 | igcewieling | why not use CALLLERID(name) and num directly in your subroutine. |
| 17:53.30 | martinvw | igcewieling: oh, DumpChan() looks nice, didn't know about that yet. |
| 17:54.07 | martinvw | igcewieling: that's what I tried first, but Dial() seems to replace the CALLERID values. Before Dial(), they contain infos about the caller, and after Dial(), they contain infos about the extension that is being dialed. |
| 17:54.42 | igcewieling | that's not the way it works. |
| 17:56.21 | martinvw | igcewieling: hm, not sure, maybe I've got something completely mixed up. But when I add same => n,NoOp(CALLERID(num) = ${CALLERID(num)}) right before my Dial() call, I get "CALLERID(num) = +49 176 ..." - that's my mobile phone number I'm calling from. |
| 17:56.22 | igcewieling | If you don't set the CallerID it should stay the same when dialing, if it didn't then callerid would only be passed to the destination if it is manually set -- not the way it works. |
| 17:56.58 | igcewieling | stop putting in extra spaces, they break stuff. So what is it after the dial? |
| 17:57.11 | martinvw | And in my crmaddcontact sub I get "CALLERID(num) = support" - that's the extension that called Dial() |
| 17:57.24 | igcewieling | is that an extra space around the = ? |
| 17:58.05 | martinvw | Yes, but I wouldn't have expected that to cause any problems... I don't use extra spaces in Set() assignments, only in NoOp(). That should only cause Asterisk to print stuff to the log? |
| 17:58.30 | igcewieling | noop and verbose won't break with extra spaces, but it is a terrible habit to get into. |
| 17:59.02 | igcewieling | until we see the dialplan output showing the issue there isn't much else to suggest./ |
| 17:59.11 | martinvw | I have a lot of shell background, I certainly won't fall into that trap when using Set() :) |
| 18:01.28 | martinvw | ok, DumpChan() is really helpful |
| 18:01.57 | igcewieling | you need to study "core show applications" and "core show functions" |
| 18:02.34 | martinvw | the mistake definitely has to be in the GoSub() call - it splits ${CALLERID(name)} into two separate variables, ARG2 and ARG3 |
| 18:02.35 | rmudgett | The U() Dial option executes the subroutine on the CALLED channel that is why you see CALLERID() change values. You are operating on a different channel. |
| 18:03.06 | martinvw | and also there's ConnectedLineIDNum and ConnectedLineIDName, which is probably what I'm looking for instead of CALLERID |
| 18:03.28 | igcewieling | rmudgett: is ${MASTER_CHANNEL(CALLERID(name))} a way to work around that |
| 18:03.50 | igcewieling | think of connected line as "called id" not "caller id" |
| 18:05.33 | rmudgett | I haven't really used MASTER_CHANNEL() so I'm not too familiar with what it does. |
| 18:05.44 | martinvw | What I'm still very curious about is how to properly use U(somesub^${some function}^${some function}) - how do I ensure that whatever ${some function} expands to is passed as a single argument, and not split at commas? |
| 18:06.20 | rmudgett | I have a blogs.asterisk.org article about the interception routines that is applicable. For some reason I cannot get to the site to post a link. |
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| 18:06.49 | martinvw | U(somesub^${CALLERID(name)) definitely seems to be very wrong, and U(somesub^"${CALLERID(name)") doesn't seem to work any better. |
| 18:07.56 | igcewieling | well the second one has quotes so that won't work |
| 18:08.25 | rmudgett | Dial(xxx,,U(context^"${CALLERID(num)"})) should pass the comma. You are using ^ as a substitute for a comma in the parameters of the U() option. |
| 18:08.38 | igcewieling | "abc" is not quoted string abc. It is a string of quote abc quote |
| 18:09.02 | igcewieling | martinvw: we still don't see a pastebin showing that. |
| 18:14.05 | martinvw | Dial() command, the [test] sub and the redacted DumpChan output - would that be enough? https://pastebin.com/XbuRGnvU |
| 18:14.15 | martinvw | or do you need the complete dialplan and asterisk output? |
| 18:14.53 | *** join/#asterisk peacetreaty (~peacetrea@static.36.148.9.5.clients.your-server.de) |
| 18:15.49 | martinvw | Putting quotes around the ${CALLERID(name)} changes nothing, DumpChan() will still print ARG1=Lastname and ARG2= Firstname (Company) |
| 18:15.54 | peacetreaty | hi, i just upgraded my linux-distribution. dahdi seems to work, pri show spans shows all my spans up, but asterisk is not answering my call. i even don't see an incomming call. does anyone know how to debug this? |
| 18:20.33 | *** join/#asterisk esteban (~esteban@unaffiliated/esteban) |
| 18:23.28 | igcewieling | try using ${MASTER_CHANNEL(CALLERID(name))} in [test] see if that gets the correct callerid name info |
| 18:25.20 | igcewieling | that way you don't need to pass it as an arg |
| 18:26.30 | igcewieling | fix the callerid line in sip.conf too callerid="support" is not valid. callerid=1234 "test" is valid, though I doubt that is the current issue. |
| 18:26.47 | igcewieling | notice the number before the name |
| 18:27.21 | rmudgett | You could set a variable with a single underscore so the called channel can see the CALLERID(name) or in this case you could use CONNECTEDLINE(name) to get the caller's name. The U() subroutine is executed by the called channel not the calling channel when the called channel answers. https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information#ManipulatingPartyIDInformation-PartyIDpropagation |
| 18:27.57 | igcewieling | <PROTECTED> |
| 18:28.48 | rmudgett | The full format is: "name" <number> |
| 18:29.21 | rmudgett | Although leaving off the quotes is also acceptable. |
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| 18:42.15 | rmudgett | This is what I was looking for earlier: http://blogs.asterisk.org/2017/03/29/dialplan-handler-routines-allow-customization/ |
| 18:46.30 | martinvw | Thanks for the advice everybody, but I'll think I'll temporarily revert my changes to our extensions.conf and call it a day for now. I'll be back :) |
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| 18:55.25 | peacetreaty | i figured it out by my self. thanks |
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| 19:27.10 | nny | Looking for some creative thinking here. I have phones that can be set to CALLFWD when someone is away. When they do, they use the same context to dial out as a normal call. As such the callerid is changed to the normal outbound CID via the dialplan. I have some rather verbose ideas on how to check channels and change CID but wondering if there's a better way |
| 19:29.19 | nny | the phones are Cisco SPA series, I am looking into how the forward is handled in the system. |
| 19:30.43 | [TK]D-Fender | CID should remain the same as the original call |
| 19:31.17 | [TK]D-Fender | <PROTECTED> |
| 19:32.37 | nny | I am working on a test setup right now, I'll do some of my own investigation and report back, thanks. |
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| 21:37.55 | Janos | hey there, question, I´m trying to build an api to be used as a realtime source for asterisk queues and queue_members. I have setup asterisk and everything seems to be in place, I get the request, but now I´m not sure what to send back to asterisk |
| 21:38.11 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 21:38.47 | Janos | according to the docs (https://wiki.asterisk.org/wiki/display/AST/cURL#cURL-multi) I have to respond with urlencoded data and thats fine but what are the names of the parameters, and which ones are required is what I don't know |
| 21:39.14 | Janos | is there any place where I can read more in depth documentation ? or should I go to the source ? |
| 21:41.13 | Janos | also is there a list where I can find out what exactly are the "things" that can be loaded with realtime ? |
| 21:58.29 | *** join/#asterisk TheGoodGuy (~quassel@mail.thegoodguysgroup.ca) |
| 21:58.43 | TheGoodGuy | Hey Guys, Maybe someone can point me in the right direction, I am looking for solution for our phone system, That basicly allows you to schedule voip calls via a web GUI anyone know of a 3rd party application / Asterisk plugin that does somthing similar? |
| 21:58.52 | TheGoodGuy | The idea is, we have our security guards making their rounds, They need to call in when they get to a site, if they do not call in, Start calling them, If they do not answer and confirm okay with a 1 and their pin or emp number, call the monitoring center.. etc etc.. |
| 22:07.33 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:08.42 | [TK]D-Fender | TheGoodGuy, Everything is 3rd party starting with you as the admin |
| 22:09.19 | [TK]D-Fender | TheGoodGuy, I'm not faminliar with any ge generically written that would serve for this as-is. |
| 22:23.54 | *** join/#asterisk sibyakin (~sibyakin@188.162.238.247) |
| 23:01.57 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 23:22.08 | *** part/#asterisk kharwell (kharwell@nat/digium/x-nikvonawtajfikct) |
| 23:40.30 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 23:55.02 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 23:55.51 | yaymuffins | Hello |
| 23:58.31 | yaymuffins | I'm trying to setup SIP for personal use, there two places I want connected to the same server, Since this is a home setup and not an enterprise, I do not have multiples public IPs and I'm behind a NAT, but there a VPN linking my home network were the SIP server is to a Linux NAS at the second place. |
| 23:59.23 | yaymuffins | Is there some sort of SIP Proxy I could use from the NAS to connect to the SIP server and then to connect the phone on the linux box? |