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| 02:01.09 | karelk | in Asterisk console, some messages are colored |
| 02:01.18 | karelk | such as the errors |
| 02:01.34 | karelk | where does this coloring come from? Can it be modified ? |
| 03:05.36 | *** join/#asterisk infobot (ibot@c-174-52-60-165.hsd1.ut.comcast.net) |
| 03:05.36 | *** topic/#asterisk is Join us at AstriCon 2019 in Atlanta! http://www.astricon.net/ -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.29.0 (2019/10/08) 16.6.0 (2019/10/08); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 03:27.13 | snuff-work | mm its shell based.. and dont recall there is an 'easy' way to change that |
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| 06:54.22 | asteriskdevdude | heya folks |
| 06:55.10 | asteriskdevdude | so i encrypted the number thats supposed to be called |
| 06:55.40 | asteriskdevdude | then decrypt in asterisk via SHELL application |
| 06:56.56 | asteriskdevdude | problem is that my encryption string is around 220 characters in length, ${EXTEN} stops at around 79 characters |
| 06:57.07 | asteriskdevdude | any solutions for this? |
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| 07:40.12 | asteriskdevdude | did i miss any replies? |
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| 08:30.18 | asteriskdevdude | keep going dced |
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| 09:41.29 | sibiria | you have a number, in e.164 or so, and you manage to inflate it to 220 characters long?! |
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| 09:42.33 | sibiria | sounds like you're encrypting a whole file, and using zip or something for it |
| 09:42.45 | sibiria | go for a stream cipher solution instead using openssl or similar |
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| 09:44.54 | sibiria | it will add only a small number of bytes for the IV and encryption rounds etc. |
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| 16:44.27 | *** join/#asterisk lagzilla (lagzilla@2600:3c03::f03c:91ff:fe6e:5ed9) |
| 16:46.09 | lagzilla | Is it possible to overwrite a SIP header, for example is I wanted to change p asserted identity. I found addheader but it looks like that does what it implies and will just insert a second header. |
| 16:48.50 | rmudgett | lagzilla: p-asserted-identity is caller-id/connected-line. You change that by changing values in the CALLERID dialplan function. Also CONNECTEDLINE |
| 16:51.06 | lagzilla | There are other fields I need to change to that was just an example, I need to add geo location to the From field too, when I send the call back out |
| 16:51.31 | rmudgett | https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information |
| 16:52.58 | lagzilla | thank you |
| 16:55.53 | Samot | Also, just FYI. Headers stack. |
| 16:56.30 | Samot | So unless you expressly remote that specific header in the stack, you don't overwrite you add. |
| 16:57.47 | lagzilla | Yeah I figured out they stack, and reading the forms it looks like asterisk doesn't support replaces outside of transfers |
| 16:58.08 | lagzilla | I'm thinking I might have to do it in an SBC |
| 16:59.09 | Samot | No. |
| 16:59.18 | Samot | PAI headers are not required headers. |
| 16:59.27 | Samot | So you would be adding them. |
| 16:59.34 | Samot | And you can add as many as you'd like. |
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| 17:00.35 | Samot | Then again, Asterisk is a B2BUA. So if you get a call from your carrier, the headers on that channel don't automatically pass to the channel you're going to dial. |
| 17:00.45 | Samot | Or vice versa. |
| 17:04.31 | lagzilla | Oh TIL |
| 17:05.11 | lagzilla | So how would I change the FROM header to go from what's set fromuser to some variable in extensions.conf |
| 17:05.12 | igcewieling | For most things. |
| 17:08.16 | Samot | What fromuser? |
| 17:09.09 | lagzilla | In sip.conf doesn;t the from value come from fromuser=something |
| 17:09.29 | Samot | Or you set the CallerID(num) |
| 17:09.57 | Samot | Chan_SIP supports modifying the from URI |
| 17:10.10 | Samot | But it really sounds like you need a proxy/SBC. |
| 17:10.16 | lagzilla | Is there any way for that to be dynamic? |
| 17:10.29 | Samot | Something meant to read the entire SIP message and manipulate it. |
| 17:11.19 | Samot | Sure. But again, with the amount of headers and probably overall message, a proxy/sbc is the better choice. |
| 17:11.49 | lagzilla | okay |
| 17:11.51 | lagzilla | thank you |
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| 19:47.26 | MLC | In the output from a DumpChan(), do the following lines indicate the codec that is in use? |
| 19:47.26 | MLC | WriteFormat= ulaw |
| 19:47.26 | MLC | ReadFormat= ulaw |
| 19:58.40 | sibiria | certainly looks like it |
| 19:58.52 | sibiria | but there are other ways to get this as a single string |
| 19:58.57 | MLC | yes |
| 19:58.59 | file | kinda. |
| 19:59.10 | sibiria | CHANNEL(audioreadformat), CHANNEL(audiowriteformat) |
| 19:59.20 | MLC | CHANNEL(audionativeformat) may be what I need. |
| 19:59.25 | file | raw is what is being sent/received with the endpoint |
| 19:59.33 | file | non-raw is what whatever is handling the channel is using |
| 20:00.16 | MLC | that makes sense. I was testing g722 with my sip provider. |
| 20:00.19 | file | soooo a Playback with only gsm sound files installed would have a writeformat of gsm, and then a rawwriteformat of whatever was negotiated (ulaw or g722 for example) |
| 20:00.35 | MLC | the non-raw was slin16 and the raw was g722 |
| 20:04.53 | MLC | the raw values don't seem to be available in the CHANNEL() function |
| 20:05.38 | file | native is them there |
| 20:05.44 | file | internally they're called raw |
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| 20:48.19 | MLC | can I influence which codec is used for an outgoing call? I listed g722 first but it always uses ulaw, even though the provider supports both. |
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| 23:52.46 | prologic | Is IncrediblePBX still maintained/recommended? I'm trying to find a newer version but I'm not seeing anhthing beyond ~2016 |
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