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| 00:27.42 | pjensen00 | Is there a way to capture the output of DumpChan in a dialplan? Or is there something similar to it that's a dialplan function? |
| 00:30.38 | [TK]D-Fender | nope |
| 00:30.47 | pjensen00 | Fudge. |
| 00:31.24 | [TK]D-Fender | You can sorta hack things by calling asterisk -rx and doing a channel dump that way and parsing it out. |
| 00:31.32 | [TK]D-Fender | messy but possibl. Next would be AMI |
| 00:31.36 | [TK]D-Fender | but nothing clean |
| 00:32.13 | pjensen00 | That's what I was afraid of. Turns out, accessing individual asterisk channel variables is expensive when each read or write is a REST request. |
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| 14:35.19 | swi_ | Hello. |
| 14:36.38 | swi_ | I'm not sure if it right place to ask. I have panasonic pbx (tde200), and i need get info (calls, caller-id) from it to custom application. Is it possible to use asterisk as some kind of proxy for that ? |
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| 14:39.41 | eschmidbauer | Is there anyway to reload rtp.conf without restarting asterisk? |
| 14:40.59 | [TK]D-Fender | swi_: Depends if your PBX uses a VoIP Protocol for its trunking and is open enough for * to sit in between |
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| 14:44.33 | swi_ | [TK]D-Fender: is there any way to check it ? |
| 14:44.49 | [TK]D-Fender | swi_: You should already know the answer to the first |
| 14:45.15 | swi_ | [TK]D-Fender: i did not install this panasonic not even programm it :) |
| 14:45.59 | [TK]D-Fender | swi_: No excuse for not know how it gets its calls |
| 14:46.09 | [TK]D-Fender | knowing* |
| 14:46.19 | [TK]D-Fender | So .... how does it get its calls? |
| 14:48.43 | swi_ | it' connected to telecom company lines to it's CO ports afaik. input lines not IP |
| 14:49.28 | swi_ | in configuration utility i see some parameter like VoIP address/port. don't know if it present in hardware though |
| 14:51.21 | [TK]D-Fender | Then for * to sit in the middle would both delay your calls (for having to see them come in on CO line for CID, and then call out), and require a card with at least the same number of FXO ports as you use now .... AND a the same number of FXS ports to send the call on to the Panasonic |
| 14:51.36 | [TK]D-Fender | This would be a crazy expense. Better off replacing int entirely |
| 14:54.59 | swi_ | i see. |
| 14:56.24 | swi_ | seems like i need than find out how to communicate with that panasonic propriertary crap :( |
| 14:56.35 | swi_ | [TK]D-Fender: thanks a lot |
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| 15:42.54 | stefan27 | is there a limit to how long strings dialplan variables can be set to ? |
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| 15:47.48 | newtonr | stefan27, i think it is 255 |
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| 15:48.46 | newtonr | mjordan, maximum character length of string channel variables? is it 255 ? |
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| 15:54.25 | mjordan | I actually think its longer, 1024 sounds right. But my memory could be wrong. |
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| 15:57.55 | stefan27 | thx |
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| 16:01.21 | stefan27 | but there is no limit to the amount of variables I can set on a channel? |
| 16:02.06 | stefan27 | I can do Set(var1=x) , Set(var2=x) , ... , Set(var10000=x) ? |
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| 16:42.58 | TandyUK | hey guys im really confused |
| 16:43.09 | TandyUK | our asterisk is logging LOTS of lines like |
| 16:43.33 | TandyUK | [2016-02-05 16:40:10] NOTICE[729]: chan_sip.c:17045 handle_request_register: Registration from '1100 <sip:1100@##EXTERNALIP##>' failed for '188.138.75.217' - No matching peer found |
| 16:43.52 | TandyUK | now ive verifird that our pbx is not accessible from the outside world, there are NO port forwards, no dmz, etc etc |
| 16:44.03 | TandyUK | so how the hell am i seeing failed login attempts from our WAN |
| 16:45.34 | newtonr | TandyUK, sounds like you probably want to wireshark that traffic :) |
| 16:45.49 | TandyUK | yeah i just dont get how this stuff is getting throug hthe firewall |
| 16:45.53 | TandyUK | port 5060 is closed, period |
| 16:46.12 | TandyUK | external tools like 'canyouseeme.org' confirm that |
| 16:46.33 | newtonr | I'd wireshark the internal traffic and see where it is coming from |
| 16:50.41 | TandyUK | hmm got one runnign now |
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| 17:57.41 | zro1 | Hi, quick question: I'm working with Asterisk 13 using PJSIP set to real time for endpoints, aor, and auths. One of the endpoints is a Cisco SPA 504G, with a 500S sidecar with a blf set to another end point. On Asterisk 13, the BLF set with call pickup functionality does not even attempt to dial the set pickup code. While on another server that runs chan_sip and asterisk 1.8, it works fine. |
| 17:58.22 | zro1 | The only difference I saw really between the sip messages (located here pastebin.com/GiKVdJpr) was taht the contact header lacked a username. |
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| 18:31.19 | Jesterboxboy | i have a question, when i use an expression in a dialplan i.e 8.25 * 1.2 asterisk gives back 9.9 but ideally i would need it with two digits after the decimal point, is that somehow possible? |
| 18:33.40 | [TK]D-Fender | * dialplan != rel programming. Use an outside script instead |
| 18:33.48 | [TK]D-Fender | real* |
| 18:35.21 | Jesterboxboy | mhm thanks, ill honor that |
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| 19:31.00 | pankid | Is there a place other than subscriptions a blf can use for monitoring an extension? Somehow sip show subscriptions is blank, but there are a bunch of blfs in use. |
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| 19:41.43 | [TK]D-Fender | proper place is "core show hints" |
| 19:45.01 | pankid | [TK]D-Fender: thanks. That is also blank though. |
| 19:45.08 | [TK]D-Fender | Show us |
| 19:46.27 | pankid | ready-174-36-204-34*CLI> core show hints |
| 19:46.27 | pankid | There are no registered dialplan hints |
| 19:46.27 | pankid | ready-174-36-204-34*CLI> |
| 19:46.35 | pankid | Sorry for the multi line |
| 19:50.32 | [TK]D-Fender | Then it looks like you didn't set anything up to be monitorable |
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| 20:54.25 | Guest41 | moo ! |
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| 21:20.37 | waj | I have a pjsip endpoint successfully registering to callcentric.com but I get a "403 Incorrect Authentication" whenever I try to dial. Any ideas? The same account with a softphone works just fine. |
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| 22:22.17 | pjensen00 | Anyone know the max length a channel variable is? |
| 22:22.29 | pjensen00 | *what the max length |
| 22:25.29 | cresl1n | First glance makes it seem like there aren't limits - at least when using pbx_builtin_setvar_helper |
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| 22:26.51 | pjensen00 | Interesting. That'd be nice for me. |
| 22:27.09 | cresl1n | obviously your heap space for the process would be a limit :-) |
| 22:27.59 | pjensen00 | Ha, I won't need THAT much but I do have to serialize all the channel variables I need in a Stasis app into a single var. |
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| 22:28.53 | cresl1n | I didn't check the stasis code path, so there might be some arbitrary limitations there that I overlooked |
| 22:29.22 | pjensen00 | I'll know soon enough if Stasis can't handle a really long variable. |
| 22:35.29 | rmudgett | The define for channel names is 80 which includes the NULL terminator. So the max length is 79. |
| 22:35.44 | cresl1n | Thanks rmudgett :-) |
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| 22:36.37 | xochilpili | hi all! |
| 22:36.40 | rmudgett | Oops. Wrong question answered. |
| 22:38.00 | rmudgett | There isn't a max channel variable value length defined. However there likely is some fixed array size still in the system. |
| 22:38.12 | cresl1n | yessssâ¦.. |
| 22:38.29 | xochilpili | i have an asterisk in : server_A and server_B, in server_A there's a trunk to my isp provider, then, in server_B i create a trunk to server_A, so, i want to make a call from server_B to server_A and then go to isp's trunk |
| 22:39.38 | xochilpili | sip show peers, said 9999/s << this is an extension in server_A which i use as a trunk in server_B |
| 22:42.33 | *** topic/#asterisk by putnopvut -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.7.2 (2016/02/05), 11.21.1 (2016/01/15); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 22:43.12 | xochilpili | so, when i call from server_B to server_A in order to use the isp's trunk, server_A said: Call from 's' (10.0.1.107:5060) to extension '98175116' rejected because extension not found in context 'default'. where '98175116' is a phone number in real world |
| 22:46.34 | xochilpili | how is it possible to server_A try to find that extension "98175116" in local dialplan if that is a phone number, i cant declare all phone numbers in dialplan, there's something im doing wrong, but i dont figured it out |
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| 23:14.50 | rmudgett | xochilpili You should read about extension patterns here: https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
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