| 00:05.14 | [TK]D-Fender | it hasn't |
| 00:06.11 | [TK]D-Fender | There are probably only minor tweaks to be done at most |
| 00:14.28 | simbalion | how do I tell if asterisk is configured to use sip or pjsip? |
| 00:15.12 | simbalion | and is there an advantage to one or the other? |
| 00:17.20 | Samot | Well.. |
| 00:17.35 | Samot | Asterisk by itself requires you to do all the configuration. |
| 00:17.50 | Samot | So, in that aspect, you'll know because you configured it. |
| 00:18.18 | Samot | If you decide to go with a GUI distro, they generally support both PJSIP and Chan_SIP if they support 13+ |
| 00:18.31 | simbalion | oh I assumed there was a default |
| 00:18.36 | Samot | And you'll be able to use both or one if you want. |
| 00:18.38 | Samot | No. |
| 00:18.42 | Samot | Asterisk is a toolkit. |
| 00:19.35 | simbalion | I followed a guide from the trunk provider for setting up the trunk configuration in sip.conf but I dunno what else I need to do |
| 00:19.41 | Samot | There are about a dozen of so channel drivers |
| 00:19.56 | Samot | From SIP to IAX to DAHDi |
| 00:20.28 | simbalion | I think I want to use SIP? |
| 00:20.54 | simbalion | The last time I worked with Asterisk was like 2010 so I'm assuming a lot has changed but I am just looking to set up a small biz pbx |
| 00:23.29 | simbalion | are channel drivers loaded in modules? |
| 00:26.04 | simbalion | hrm I think I am getting the hang of it |
| 00:26.13 | simbalion | I might write a tutorial for my use-case when I'm done |
| 00:29.29 | simbalion | yay I got chan_sip loaded.. had a syntax error in the conf... now to get linphone working |
| 00:42.04 | simbalion | is there a limit to secret length? 'sip show users' shows it truncated |
| 00:54.11 | *** join/#asterisk pkx2 (~pkx@unaffiliated/pkx) |
| 01:11.11 | simbalion | well, at least the trunk is connecting alright, so I'm like 20% of the way heh.. farther than I expected really |
| 01:19.35 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:19.35 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.2 (2018/02/21), Standard: 15.2.2 (2018/02/21); 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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| 15:54.53 | Dovid | Is there any option to readexten that does not need a #? |
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| 16:16.32 | Samot | igcewieling: You use externnotify? |
| 16:18.23 | igcewieling | Samot: no. |
| 16:18.33 | Samot | Damn. |
| 16:18.36 | igcewieling | I use externpasscheck |
| 16:18.41 | Samot | Yeah, me too. |
| 16:18.44 | Samot | That works fine. |
| 16:19.06 | Samot | It just doesn't seem like externnotify is being called when a voicemail is left or the box is checked. |
| 16:19.22 | Samot | I'm trying to figure out where to look to see if it is called. |
| 16:22.48 | Samot | Oh. |
| 16:22.55 | Samot | I see it |
| 16:25.39 | [TK]D-Fender | Dovid, if you set a max length |
| 16:25.50 | [TK]D-Fender | Dovid, the instructions say it all |
| 16:25.59 | Dovid | TK: I didnt see that. Let me look agian |
| 16:26.04 | Dovid | agaun* |
| 16:26.06 | Dovid | again* |
| 16:26.06 | [TK]D-Fender | X-Rob, thanks for the yum advice last night BTW. All good now. |
| 16:26.20 | Samot | And found the issue. |
| 16:28.06 | Dovid | TK: Where do you see that here? https://pastebin.com/GZH4TczF |
| 16:32.09 | Samot | Reads a '#' terminated string of digits from the user into the given variable. <-- Think that answers your question. |
| 16:32.38 | Samot | Since it's whole thing is a "# terminated string" then not using # means it's not the same type of string. |
| 16:32.43 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Read |
| 16:32.51 | Samot | Read is not ReadExten |
| 16:32.51 | [TK]D-Fender | READ. Not READEXTEN |
| 16:33.24 | [TK]D-Fender | I don't see the point of ReadExten so far... |
| 16:33.32 | Samot | Well, the question was about ReadExten |
| 16:33.36 | Samot | So it's answered. |
| 16:33.42 | Samot | The better option is probably Read |
| 16:34.05 | [TK]D-Fender | Read was always the app I've used since pre 1.0 |
| 16:34.23 | [TK]D-Fender | Don't know when ReadExten happened.... and it only does less. |
| 16:34.35 | Dovid | TK: I will try read |
| 16:43.23 | Samot | Yeah, I just use Read() when needed. |
| 16:51.36 | Dovid | TK: So there is no option like stream_file where I select what digits I want? Like if I want the options to only be 1 or 2? |
| 16:52.47 | [TK]D-Fender | WRite it out as fuller dialplan yourself at that point |
| 16:52.56 | [TK]D-Fender | You can see the application list like everyone else |
| 16:53.01 | [TK]D-Fender | the answer isn't hidden |
| 16:55.14 | Dovid | TK: I thought so I just figured I would ask.. I can write out the dial plan but it's dynamic. Some times there are two options sometimes there are 5 |
| 16:56.33 | [TK]D-Fender | What kind of "options? |
| 16:56.50 | igcewieling | and what will cause them to vary in number? |
| 17:04.59 | Dovid | TK: Languague selection. Some times 1 is english and 2 is french. Some times 1 is french, 2 is english and 3 is russian |
| 17:07.07 | igcewieling | Write an IVR menu for each, like everyone else. |
| 17:07.33 | *** join/#asterisk razzex (~razzex@185.24.236.142) |
| 17:09.13 | razzex | Is there any way to analyze incoming sip messages? To be specific, i'd like to see the "content-lenght" value of a message. I know that chan_sip has implemented methods in its sourcode for this, but i'd like to do it though an external module. Is it possible? |
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| 17:18.35 | file | you'll have to be more specific... in chan_sip it's not extendable so you can't |
| 17:18.45 | file | chan_pjsip has a modular design with APIs, so you can insert yourself in different places |
| 17:20.04 | razzex | Sorry im quite a beginner - does that mean that an external module can use chan_pjsip's functions? |
| 17:20.46 | igcewieling | razzex: are you a programmer? |
| 17:20.48 | file | an external .c module for Asterisk can use the APIs we provide for PJSIP to insert itself and do various things |
| 17:21.28 | razzex | I see. So no way to do it with chan_sip. Okay, thanks alot. |
| 17:22.21 | file | chan_sip provides no APIs or ways to extend/inject logic |
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| 18:11.24 | razzex | Could anyone please recommend me where to start if i want to do a custom pjsip module which can analyze specific fields in sip header? |
| 18:13.51 | file | there is no guide, best case is to examine the available APIs and look at an existing module that does close to what you want |
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