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| 00:22.24 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0-rc3 (2018/09/20), Standard: 15.6.1 (2018/09/20); 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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| 14:05.16 | igcewieling | Some days I really hate VoIP. |
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| 14:19.17 | jayvines | ice - what happened? why the hate? |
| 14:19.27 | jayvines | sorry *igc |
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| 14:23.19 | jayvines | Does anyone have any good recommendations for spanish language elastix support? a friend doesn't speak english and needs some help at work |
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| 14:25.57 | igcewieling | jayvines: for no apparent reason ringback stopped working sometime this weekend or this morning. |
| 14:28.46 | jayvines | well, that stinks. but as you know, there must be a reason |
| 14:29.02 | jayvines | so, what did someone change and forget about or not tell you about? |
| 14:31.21 | igcewieling | I'm the only one who can make changes and I don't upgrade all the servers at once. The workaround is to Answer() then Ringing, but that is an ugly hack. As far as I can tell from the addled brains of customers the issue isn't with every customer. |
| 14:35.19 | jayvines | that's weird, dude. something changed. you made a change that you thought was harmless and was actually a dependency. for sure |
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| 17:15.20 | igcewieling | Does anyone know why I might get these errors when a parallel dial is answered. i.e. Dial(SIP/1234&SIP/5678&SIP/9824) and 1234 answers, I get that message for the other extensions in the Dial. [2018-10-01 13:13:58] WARNING[3042][C-00000107]: chan_sip.c:24072 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '64db55c74c557f8101c957900973e31b@10.0.0.254:5060'. Giving up. |
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| 17:16.51 | gedia | hello all. quick question because I have been reading things online that imply the opposite: is it possible during SDP negotation between two SIP UAs for them to use different codecs when they send audio in a media stream? Or they always have to agree on a common codec? |
| 17:19.51 | file | are you asking in regards to SDP in general? |
| 17:22.01 | igcewieling | gedia: SIP UAs don't usually talk to each other, they talk to Asterisk which can transcode. |
| 17:23.11 | Samot | gedia: Unless there is some B2BUA (like Asterisk) or SBC/Media server that handles the media and can transcode...yes the to SIP endpoints must talk on the same codec. |
| 17:23.21 | Samot | s/to/two/ |
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| 17:29.55 | jameswf | ?botsnack |
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| 17:31.46 | [TK]D-Fender | ~areyouadog ? |
| 17:31.46 | infobot | Bark! Bark! |
| 17:31.50 | [TK]D-Fender | ~botsnack |
| 17:31.50 | infobot | thanks, [TK]D-Fender |
| 17:32.01 | [TK]D-Fender | ~infobot |
| 17:32.01 | infobot | methinks infobot is A program on the IRC that helps users, ask it to do something by putting a ~ and then say a command! |
| 17:36.12 | jameswf | ~botabuse |
| 17:36.12 | infobot | hmm... botabuse is fun |
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| 19:33.09 | pa | hi |
| 19:33.56 | pa | i have an extension that looks like: exten => _XX.,1, |
| 19:34.00 | pa | etcetera |
| 19:34.03 | pa | i was wondering |
| 19:34.11 | pa | is there a way i can prevent dialing certain number? |
| 19:34.39 | pa | i just discovered i misconfigured my firewall and someone called a number of african phone lottery numbers using my asterisk.. |
| 19:35.05 | pa | i hope i fixed it, but in any case, could i somehow whitelist or blacklist the dialed extension? |
| 19:38.52 | [TK]D-Fender | make a match those other numbers & patterns, or check for them in that exten |
| 19:39.35 | [TK]D-Fender | or come up with a better pattern than "anything 3 digits or longer" for your outbound dialing |
| 19:39.52 | [TK]D-Fender | That lets just about anything through. |
| 19:43.00 | *** part/#asterisk foo (~foo@unaffiliated/foo) |
| 19:44.35 | pa | [TK]D-Fender: yes that's the point: everything right now goes through that |
| 19:45.08 | [TK]D-Fender | Stop using psychotically open patterns like that would be a good start |
| 19:45.25 | pa | well atm if i want to call a number i have to go through that extension |
| 19:45.30 | pa | it's my ISDN path |
| 19:45.36 | [TK]D-Fender | Why would you let them dial 5000 out? Where is that possibly valid? |
| 19:45.54 | pa | well it's invalid but why would someone dial it? |
| 19:46.39 | pa | normally i match specific numbers earlier |
| 19:46.47 | pa | and this is the catchall that should go out |
| 19:51.20 | [TK]D-Fender | Where are your callers expected to be calling out to? |
| 19:53.49 | pa | [TK]D-Fender: well up till now technically wherever they wanted. Now i would like to put some restrictions |
| 19:54.18 | [TK]D-Fender | that's where you ALLOW them right now |
| 19:54.23 | [TK]D-Fender | I asked where you EXPECT themt o be calling |
| 19:58.16 | pa | [TK]D-Fender: ok, so lets say only national calls or two or three specific international prefixes |
| 19:58.52 | [TK]D-Fender | then make those |
| 19:59.55 | pa | [TK]D-Fender: hm, right. Is there a way to make a template extension, and just change the match? |
| 20:00.20 | [TK]D-Fender | just make the matches you need |
| 20:00.20 | pa | like in my case i have 9-10 steps, that i would otherwise copy paste across all 4 extensions |
| 20:00.31 | [TK]D-Fender | then make macros |
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| 20:04.43 | [TK]D-Fender | packs up to head home |
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