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| 00:23.25 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.0 (2018/05/01), Standard: 15.4.0 (2018/05/01); 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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| 13:57.49 | [sID] | hi, how to cut this value "616665654" "<-- this |
| 13:58.02 | [sID] | in dialplan |
| 13:59.23 | [TK]D-Fender | Depends what you actually want to do. |
| 14:03.26 | [sID] | have the same number without " |
| 14:04.38 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_STRREPLACE |
| 14:06.07 | [sID] | Ok, I've done it. thanks |
| 14:07.57 | igcewieling | CUT might work too. |
| 14:08.41 | [TK]D-Fender | Depends what the consistency looks like |
| 14:08.50 | igcewieling | as would the standard variable stuff something like ${MYVAR:1:01} |
| 14:09.04 | igcewieling | ${MYVAR:1:-1} |
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| 14:35.07 | igcewieling | What might cause linkedid to be truncated to 32 chars in the CDRs, even after I altered the table to make linkedid varchar(64) and restarted asterisk? |
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| 14:39.06 | Samot | Being able to accept 64 characters doesn't mean 64 characters was submitted. |
| 14:41.50 | *** join/#asterisk Guest9201 (b92009fa@gateway/web/freenode/ip.185.32.9.250) |
| 14:42.01 | Samot | I don't think the linkedid is ever over 32 characters. |
| 14:42.01 | Guest9201 | Hi, do I really need the hangup in the foobar exten? https://pastebin.com/8RKWZYcL |
| 14:42.31 | Samot | tIt would be strange to have a database spec with less characters than what the data submitted would always be. |
| 14:49.44 | igcewieling | Samot: I guess now I have to ask does anyone know how to make Asterisk log more than 32 chars for a linkedid in the CDRs. LinkedID in CLI: daffy-01.nyigc.net-1526481871.166198 LinkedID in CDR=daffy-01.nyigc.net-1526481871.1 |
| 14:51.30 | Samot | Is that how it appears in the log? |
| 14:51.38 | Samot | Or how it appears in the actual database? |
| 14:55.34 | igcewieling | yes. |
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| 14:56.02 | igcewieling | the CLI is from the log file and the CDR one is from the databasr. |
| 14:56.59 | Samot | And this was after you adjusted the length? |
| 14:57.41 | igcewieling | yes, that was 3 mins ago when I pasted the info. |
| 14:59.21 | Samot | And if you create a new column with a new name and submit the value there, does it truncate? |
| 14:59.25 | Samot | I'd try that |
| 15:01.37 | file | older versions didn't have a common definition for the length |
| 15:02.23 | file | as of 13 everything uses the same length limit, and it's calculated based on other things (such as system name length, timestamp, yada) |
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| 15:04.25 | Samot | So CDR(linkedid) is going to assume it's 32 characters? |
| 15:05.54 | file | in 11 it would |
| 15:06.02 | file | 13 is... math... 150+ |
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| 15:06.13 | igcewieling | I'm writing a script to link each CDR record we get from the carrier to the Asterisk CDR by linkedid and occasionally get duplicated linkedids on unrelated calls.. |
| 15:08.19 | igcewieling | file, thanks for the info |
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| 15:20.03 | Guest9201 | Something that really grind my gears is that HANGUPCASE is set to 0 upon receiving a SIP CANCEL. And the hangup app cannot handle 0 |
| 15:21.37 | Samot | When is the CANCEL being sent? |
| 15:22.38 | Samot | Is it being sent during setup or after an active call hangsup? |
| 15:22.46 | Samot | By active, I mean answered. |
| 15:24.28 | Guest9201 | On setup |
| 15:24.42 | Samot | And what are they sending a CANCEL back for? |
| 15:25.01 | Samot | Is this an incoming call from a provider or an outgoing call? |
| 15:25.22 | Guest9201 | It is an outgoing call, and the CANCEL is sent to CANCEL the call setup? I persume. |
| 15:26.08 | Samot | No. |
| 15:26.20 | Samot | If Asterisk is making this call, Asterisk should send the CANCEL |
| 15:26.28 | Samot | They should send back proper replies to the INVITE |
| 15:27.11 | Samot | CANCEL is a request not a reply. |
| 15:27.17 | Samot | It should never be sent as a reply. |
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| 15:28.49 | Guest9201 | We're sending back request terminated upon received cancellation |
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| 15:29.32 | Samot | You are sending an INVITE |
| 15:29.41 | Samot | That is a request. It should only get valid replies. |
| 15:29.47 | Samot | A CANCEL is not a valid reply. |
| 15:30.02 | Guest9201 | We're sending both the invite and the cancel |
| 15:30.09 | Samot | OK. |
| 15:30.16 | Samot | So you cancelled the call |
| 15:30.19 | Guest9201 | Sure |
| 15:30.33 | Samot | Is it because the Dial() timer was exceeded? |
| 15:30.58 | Guest9201 | No, the caller did not want to wait any longer and had better stuff to do |
| 15:31.30 | Guest9201 | *originator |
| 15:32.11 | Samot | OK, so the HANGUPCAUSE is a code that is mapped to a SIP Reply Code. |
| 15:32.19 | Samot | CANCEL does not have a SIP Reply Code. |
| 15:32.28 | Samot | Outside of 200 OK or ACK |
| 15:32.40 | Samot | So there is nothing to map so you're going to get a zero |
| 15:33.30 | Guest9201 | So how do Asterisk handle this scenario? Now it looks like it returns Request Terminated on the CANCEL, back to the originator. |
| 15:33.30 | Samot | You can use hangup handlers on the caller's channel to deal with that so you can send whatever you need or set whatever you need for your app. |
| 15:34.13 | Samot | That is the ACK/OK reply to the CANCEL |
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| 15:36.05 | *** mode/#asterisk [+o gtjoseph] by ChanServ |
| 15:37.15 | Guest9201 | I will add the hangup handler for the CANCEL event |
| 15:37.52 | Samot | Well.. |
| 15:38.17 | Samot | The hangup handler will fire no matter what when the channel is dropped |
| 15:38.24 | Samot | Hits h extension. |
| 15:38.39 | Guest9201 | same => n,gotoif($[${HANGUPCAUSE} = 0]?result_CANCEL,1) |
| 15:38.44 | Guest9201 | we just want some control over them |
| 15:38.48 | Samot | That's fine. |
| 15:38.51 | Samot | You can do that |
| 15:39.02 | Samot | Just remember though, 0 = No Mapped Response. |
| 15:39.56 | Samot | So 0 isn't always going to be the result for this scenario. |
| 15:40.41 | Samot | It could be the result for others where Asterisk doesn't have a causecode mapped to a SIP code. |
| 15:41.58 | Guest9201 | I think that is fine |
| 15:42.41 | Samot | Just letting you know |
| 15:42.59 | Samot | But that dialplan line is fine. |
| 15:43.27 | Samot | It can run to handle the Dial() now, hangup handlers can be assigned to the channels... |
| 15:43.53 | Samot | So you could execute different things on each channel when they hit the h extension. |
| 15:48.26 | Guest9201 | My mappings are broken, it seems like. |
| 15:48.39 | Guest9201 | On 504 Gateway Timeout we replied 603, and on 603 we replied 403 |
| 15:48.45 | Guest9201 | :-( |
| 15:49.52 | Samot | I think you are looking at this wrong. |
| 15:50.10 | Samot | Asterisk is a B2BUA. There are always two channels involved in a single call like this. |
| 15:50.22 | Guest9201 | Sure |
| 15:50.28 | Samot | One from the endpoint to Asterisk and one from Asterisk to the Provider. |
| 15:50.35 | Samot | The provider is telling Asterisk 504. |
| 15:50.46 | Samot | Asterisk is telling the device 603, it was declined. |
| 15:50.59 | Samot | It's not passing the reply from the provider to the endpoint. |
| 15:51.09 | Samot | It's generating it's own reply for the endpoints request. |
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| 15:51.55 | Samot | And when the provider sends s 603, Asterisk sends a 403 to say "forbidden/not allowed" |
| 15:52.01 | Guest9201 | Yeah sure, but why would Asterisk translate 504 to 603? What is that decision based on? |
| 15:52.14 | Samot | The point of view. |
| 15:52.22 | Samot | To the endpoint it didn't get a 504. |
| 15:52.32 | Samot | Asterisk told it, "I declined the call" |
| 15:52.58 | Samot | 603 Decline means "I attempted this call, it failed and there are not other options to send it to" |
| 15:53.17 | Samot | Roughly. |
| 15:53.43 | Samot | Between the provider and Asterisk, Asterisk is the "user" in this case. |
| 15:53.56 | Samot | Between the endpoint and Asterisk, the endpoint is the "user" |
| 15:54.28 | Samot | The only time the endpoint would get a 504 is when the server (Asterisk) timesout. |
| 15:54.31 | Samot | It's not |
| 15:57.19 | Guest9201 | What is we were to provide only SIP trunks? Meaning no users never connect to this asterisk only other gateways. |
| 15:57.31 | Guest9201 | Wouldnt it make sense then to relay the correct responses? |
| 15:59.56 | [TK]D-Fender | * is not a "relay" |
| 16:00.10 | [TK]D-Fender | or proxy |
| 16:01.01 | Samot | ^^^^ |
| 16:01.10 | Samot | 11:50:10 AM <Samot> Asterisk is a B2BUA. |
| 16:01.33 | Guest9201 | Sure, but what I meant what sending the response codes over the bridge |
| 16:01.43 | Guest9201 | not translating them into what Asterisk think is correct |
| 16:02.03 | Samot | They are two channels. |
| 16:02.09 | Samot | One channel has to leave the bridge. |
| 16:02.17 | Samot | Then the other channel may or may not hangup |
| 16:02.34 | Samot | Either way, it's two channels handling calls in two different directions. |
| 16:03.03 | Samot | One side is terminating a call to asterisk and asterisk is originating a call to a destination. |
| 16:03.44 | Samot | Because what if you try Provider A and they give you a 504 and then you tell Asterisk to call Provider B |
| 16:04.00 | Samot | The endpoint is still sitting there waiting for this call to complete. |
| 16:04.14 | Samot | blindly passing back the response would make the endpoint drop the call. |
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| 16:06.11 | Samot | Remember, the endpoint sends the call to Asterisk. Asterisk processes that call, you're telling it in the dialplan to pick up a new channel and Dial(). |
| 16:11.50 | Guest9201 | Ok, you convinced me. |
| 16:36.07 | *** join/#asterisk Givemelove (0c152506@gateway/web/freenode/ip.12.21.37.6) |
| 16:37.31 | Givemelove | @Craigify Thanks for all your help yesterday. The GCC flags worked out and I was able to rebuild to 1.8 (1.6 still gave me issues). |
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| 17:29.12 | craigify | Givemelove, glad to hear it. |
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| 17:50.23 | pchero | Hello, |
| 17:50.23 | pchero | I've just made some small project related to the Asterisk. |
| 17:50.23 | pchero | I designed this for call center solution. |
| 17:50.23 | pchero | It would be really nice if you look around the project and give some stars to the project. :) |
| 17:50.23 | pchero | It's not really done yet, but some features are working. ;) |
| 17:50.23 | pchero | So, if you need some open source project related with Asterisk, this might be helpful. |
| 17:50.23 | pchero | Any question or opinion would be welcome. :) |
| 17:50.24 | pchero | https://github.com/pchero/jade |
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| 18:01.49 | craigify | pretty cool pchero |
| 18:02.19 | pchero | craigify: Thanks! Do you like it? :) |
| 18:03.22 | pchero | With 2 jade-me logged in client, you can test a call to each other. :) |
| 18:05.15 | craigify | you wrote your admin backend in C? |
| 18:05.46 | *** join/#asterisk masoudd (~masoudd@5.52.192.247) |
| 18:05.49 | pchero | Yes. :) |
| 18:06.09 | pchero | jade is backend. It written in c |
| 18:06.31 | pchero | But the other frontends are written in Angular. |
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| 19:28.51 | *** join/#asterisk corretico (~corretico@200.91.143.34) |
| 19:29.01 | corretico | hello guys |
| 19:29.11 | corretico | I need some assistance |
| 19:29.27 | corretico | I'm trying to use Cisco Phones 3905 with asterisk |
| 19:30.11 | corretico | I read some info about this phones and asterisk, but I'm confused |
| 19:30.55 | corretico | someone talks about 3 files... dialplan.xml, SEP.cnf.xml and firmware |
| 19:32.07 | corretico | I have a SEPMAC.cnf.xml.... but this file is generic for every cisco phone? |
| 19:33.25 | corretico | generic for all Cisco Phone... sorry ;-) |
| 19:35.16 | [TK]D-Fender | no it's obviously specific to each phone |
| 19:35.56 | corretico | oki |
| 19:36.37 | corretico | there is some template applicable for all? |
| 19:37.36 | [TK]D-Fender | www.cisco.com <-------------- |
| 19:39.12 | igcewieling | Wow! People love the password "1234". |
| 19:39.39 | igcewieling | Like 2/3 of the voicemail password changes are attempts to change it to 1234 |
| 19:39.55 | malcolmd | "1-2-3-4-5? That's the stupidest combination I've ever heard of in my life! That's the kinda thing an idiot would have on his luggage!" |
| 19:40.39 | craigify | hahaha nice spaceballs reference |
| 19:41.14 | malcolmd | may the schwartz be with you |
| 19:41.47 | craigify | SPACEBALLS! the flame thrower... |
| 19:41.56 | malcolmd | kids love these things |
| 19:44.04 | igcewieling | what might cause the linkedid and uniqueid to be different in a single CDR record? |
| 19:48.14 | malcolmd | one example is the "core attended transfer to channel" from https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification |
| 19:57.34 | igcewieling | thanks |
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| 22:55.33 | rpifan | hi |
| 22:57.13 | rpifan | does asterisk have python support |
| 22:57.24 | [TK]D-Fender | As in? |
| 22:57.32 | [TK]D-Fender | that question has no context |
| 22:58.20 | rpifan | oh hey [TK]D-Fender how r u |
| 22:58.24 | rpifan | i see your still alive and ar ound |
| 22:58.45 | rpifan | hows your existance going |
| 22:59.27 | [TK]D-Fender | existing |
| 22:59.30 | Samot | rpifan: Asterisk has Python support in the sense you can write scripts/AGI/AMI commands with it. |
| 22:59.48 | Samot | And then call on them, like most other programs. |
| 23:00.06 | rpifan | so not like directly connected to asterisk via some api |
| 23:00.20 | Samot | Python is a scripting language. |
| 23:00.27 | Samot | Like PHP |
| 23:00.42 | [TK]D-Fender | And "directly connected" is SUPER vague |
| 23:01.02 | [TK]D-Fender | You've been around a long time .. far long enough to know you should be specific about your goals. |
| 23:01.13 | [TK]D-Fender | And you should have an awareness of *'s configuration and interfaces |
| 23:01.25 | rpifan | i do |
| 23:01.33 | rpifan | im just asking cause iv ebeen doing python work at work |
| 23:01.37 | rpifan | and just thinking about the future |
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