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| 01:19.41 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); 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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| 05:07.41 | adonaros | good day, stupid Q, someone can pm me the answer if they like. Just hint to me, what is one way that you could add in your "outbound" context your custoemrs DIDs so that if one CX calls another CX you dont have to actually terminate the call to the PSTN and take a shortcut.. |
| 05:08.35 | adonaros | i have tried "include => inbound" but futher RTFMing, this doesnt work, cos include only kicks in if there are no matches in the current context, and outbound has _X. so therefor it matches and never tests/checks 'inbound'. |
| 05:08.54 | [TK]D-Fender | Change the order of your includes |
| 05:09.00 | [TK]D-Fender | because order matters |
| 05:09.58 | adonaros | that was a good hint. before asking more stupid questions, i will try what you said, like a good person. and report back (likely) how smart you are. |
| 05:10.02 | adonaros | good day! (: |
| 05:14.53 | [TK]D-Fender | go for it |
| 05:15.38 | [TK]D-Fender | it's usually a good idea to use "container" contexts for key things that ONLY have includes so you can really gaurantee the outcomes and not have a global override stuck in the base. |
| 05:16.37 | adonaros | wow. |
| 05:16.41 | adonaros | you were right |
| 05:17.00 | adonaros | all this time i had just included inbound in outbound along with its patterns |
| 05:17.23 | adonaros | now you can say like "no shit i was right" -- anyway, Youre smart. Thank you. |
| 05:18.01 | [TK]D-Fender | I knew one thing you we'ren't aware of. That's knowledge, not "smarts". |
| 05:18.06 | [TK]D-Fender | Happy to pass it on |
| 05:18.23 | adonaros | i will hang out in this channel more, perhaps i can help someone too. |
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| 08:34.38 | bittis | hey everyone, random question, what happens when we call mixmonitor more than once on a channel? |
| 08:34.55 | bittis | do any subsequent calls get ignored? |
| 08:36.11 | elcontrastador | I'm cutting over a client TONIGHT from ccm to Asterisk 15. Just noticed that they have about 30 faxes that were previously running ATAs with sccp. Ive replaced the ATAs with grandstream sip ATAs. Issue I see is that they have the same fax DID mapped to up to 4 faxes. I guess they want each fax going to all 4 faxes. If I list them all in a Dial with "&", it will just send open the channel to the first one that answers right? Is there anyway to |
| 08:36.11 | elcontrastador | <PROTECTED> |
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| 09:21.55 | elcontrastador | I will NOT be ignored! I demand an answer this minute! We had a false nuclear attack responded to in only 38minutes. Why is this taking longer than that? haha, jk. Too much stress and caffeine. I think I'm funny when i over caffeinated :-) |
| 09:38.45 | DanQuinney | can confirm, it's just you that thinks that elcontrastador ;) |
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| 09:38.56 | elcontrastador | lol... |
| 10:21.28 | bittis | hmm |
| 10:21.31 | bittis | well |
| 10:21.36 | bittis | i can give you an answer |
| 10:21.46 | bittis | but it will be 99.9% wrong as i don't have one |
| 10:22.02 | bittis | if any response is ok though, then sure, i agree :P |
| 10:42.45 | bittis | also suspecting nobody in |
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| 12:00.56 | elcontrastador | bittis: was the 99.9% wrong answer intended for me? |
| 12:02.55 | elcontrastador | Did any else respond while I was offline? I'm going to repost... |
| 12:03.09 | elcontrastador | I'm cutting over a client TONIGHT from ccm to Asterisk 15. Just noticed that they have about 30 faxes that were previously running ATAs with sccp. Ive replaced the ATAs with grandstream sip ATAs. Issue I see is that they have the same fax DID mapped to up to 4 faxes. I guess they want each fax going to all 4 faxes. If I list them all in a Dial with "&", it will just send open the channel to the first one that answers right? Is there anyway to |
| 12:03.16 | elcontrastador | make it send to all 4 at once? |
| 12:06.28 | file | you can't, fax is a two way negotiation |
| 12:06.55 | file | so even if you could connect them all together they wouldn't be able to negotiate and the call would fail |
| 12:07.49 | elcontrastador | ok...interesting...how were they doing it from sccp? it's configured in the ccm this way...is it possible that the fax just came in on the first that answered? |
| 12:08.15 | file | yes, it's possible it did just that |
| 12:09.45 | elcontrastador | ok...so to replicate, i'll just Dial with '&' chain...just free for all it |
| 12:09.56 | elcontrastador | sound right to u? |
| 12:10.20 | file | yes |
| 12:15.20 | elcontrastador | @file: Thanks so much! oh...one last thing! Do I need to do anything outside the default res_fax config to have fax working thru sip? |
| 12:15.40 | file | res_fax is not used if you are referring to T.38 passthrough |
| 12:16.42 | elcontrastador | ok...the module is res_fax is loaded....i was hoping shit'd just work but u know this fax thing and my deadline is looming :-)_ |
| 12:16.57 | elcontrastador | so should i just leave it or unload it |
| 12:17.07 | file | it won't hurt things... |
| 12:17.11 | elcontrastador | ok perfecct |
| 12:17.31 | elcontrastador | ...and my keyboard is going out on my dongle book pro |
| 12:17.58 | elcontrastador | what a night...fire alarm went off too...cops and fire dept....drama...i appreciate the help |
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| 13:31.43 | guivie | Hi all! How can I dial in a extension using another sip host with authentication ? something like exten => 1907,1,NoOp(SIP/${EXTEN}:user:pass@192.168.39.15) |
| 13:32.34 | guivie | Mayble something like Dial(SIP/${EXTEN}@user:pass:192.168.39.15) |
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| 13:34.58 | Samot | https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample |
| 13:35.20 | Samot | ^^ All the Dial() options for Chan_SIP are shown in the sample config. |
| 13:35.36 | Samot | Including what you are looking to do. |
| 13:36.09 | guivie | Thank Smot.. I will try that |
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| 14:06.59 | jeffspeff | I'm planning to make a db for storing configuration parameters for polycom phones that will be used by a script to generate the provisioning files. I can't decide if I should structure the tables based on firmware version or phone model. Any suggestions or advice? Thanks. |
| 14:22.58 | Samot | Well both. |
| 14:27.50 | jeffspeff | by structure the tables, i'm not referring to the columns within the tables, i'm referring to the tables themselves. should i have different tables for each model number and then have columns for the compatible configs for that model or should i have different tables for each firmware version with columns for the compatible configs and a column for the phone models supported by that firmware? |
| 14:28.20 | jeffspeff | or am i overcomplicating this? |
| 14:30.02 | Samot | Well, certain models dont have certain features. |
| 14:30.43 | Samot | VVX411 doesnt do video so having all the video settings is a waste in the config |
| 14:31.10 | Samot | Also certain features may not exist in certain firmware releases. |
| 14:31.24 | Samot | Or settings change/added/removed |
| 14:31.55 | Samot | But all the Polycom VVX phones have the same config setup |
| 14:32.07 | Samot | Same settings, same layout |
| 14:32.23 | Samot | It's how most phones work.. |
| 14:32.48 | Samot | There is a core template all the models use for that firmware and then certain settings are in templates with phones that have more options. |
| 14:33.24 | Samot | If you use a VVX500 template, that does support video, on a VVX411 the 411 will accept the config and ignore all the video pieces. |
| 14:33.54 | jeffspeff | i was leaning towards tables for each firmware version with columns for the supported features (that will be distinct for the phones) of that firmware version. |
| 14:34.11 | Samot | Well it also depends on the phone. |
| 14:34.20 | Samot | Polycoms use multiple files |
| 14:34.23 | jeffspeff | the values will be plugged into the generic templates. |
| 14:34.27 | Samot | Some phones use a single file. |
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| 16:50.13 | imcdona | something changed after Asterisk 13.17.1 that broke res_corosync. I've installed the latest version 13.19.0 and corosync is still broken. This is especially frustrating seeing as how the corosync module was recently fixed after being broke for 2 years only to be broken again after a couple builds. In order to pinpoint what build broke corosync I need to download builds after 13.17.1 such as |
| 16:50.13 | imcdona | 13.17.2 etc. What is the syntax to download a specific build of Asterisk? |
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| 17:02.40 | elcontrastador | I'm getting this nasty audio translator error that is only happening occasionally (every few mins)...first morning after cutover from ccm...very stressed...any help appreciated : https://gist.github.com/elcontrastador/09deeec66f7802b3bacfc1e468555cf4 |
| 17:03.40 | [TK]D-Fender | "one capability supports no formats"one side isn't offering any codecs |
| 17:03.48 | [TK]D-Fender | show the actual call attempt with SIP debug enabled |
| 17:05.17 | elcontrastador | yeah...very busy box...log doesn't have context as to which call is doing this...i'm getting your deb |
| 17:06.10 | jeffspeff | elcontrastador, one tool which may help you find the culprit is sngrep. |
| 17:06.31 | elcontrastador | is that within asterisk? |
| 17:07.03 | jeffspeff | no it's a 3rd party linux program |
| 17:07.56 | jeffspeff | https://github.com/irontec/sngrep/wiki https://github.com/irontec/sngrep/wiki/Installing-Binaries |
| 17:07.59 | elcontrastador | also getting this: [Jan 16 09:07:39] WARNING[17025][C-0000066b]: app_voicemail.c:8892 play_message: Playback of message /var/spool/asterisk/voicemail/hjuhsd/6013/INBOX/msg0000 failed |
| 17:08.13 | elcontrastador | never had either of this... |
| 17:08.22 | elcontrastador | "playback's a bitch" |
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| 17:18.09 | elcontrastador | [TK]D-Fender: i'm handling another emerg...i'll get back to this in a few mins (asap) |
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| 17:18.49 | elcontrastador | jeffspeff: thx. I'm going to clone it and try it in few |
| 17:19.06 | jeffspeff | or just install the binary |
| 17:19.42 | [TK]D-Fender | Sounds like Gentoo.... |
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| 17:30.49 | Zanelos | When setting up a PBX with a SIP proxy, with multiple upstream ITSP's... does the PBX choose the destination trunk signalling endpoint IP, or does the proxy? And when working with the ITSP for a new trunk setup, would you provide the external IP's of the PBX's in addition to those of any proxies in use? Or just the proxy IP's? |
| 17:36.16 | [TK]D-Fender | When you call out using a proxy then you send it to them with the destination set as the host for the itsp in question |
| 17:43.03 | kunwon1 | https://twitter.com/LevineJonathan/status/953310521775788032 |
| 17:43.08 | kunwon1 | sorry wrong channel |
| 17:43.39 | Samot | Yeah and not that big of a deal. |
| 17:43.51 | Samot | Ask anyone who's worn glasses for most their life. |
| 17:43.58 | kunwon1 | really, really wrong channel |
| 17:44.19 | Samot | Well if this was some sort of dig at Hatch, who I'm not a fan of, it really isn't a dig. |
| 17:44.26 | kunwon1 | goddammit |
| 17:44.38 | kunwon1 | i need to have some kind of a 'do you really want to paste this' dialog |
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| 20:26.31 | znoteer_ | in the output of "iax2 show registry", the column "perceived" should be the public IP address of my local machine, correct? |
| 20:29.01 | znoteer_ | it's what the peer sees us as, yes? |
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| 22:37.04 | kfife | Is there a channel variable for the automixmon filename? e.g. features.conf *3 |
| 22:40.34 | kfife | The Mixmonitor application has ${MIXMONITOR_FILENAME}, but that's null when invoked by a feature |
| 22:42.35 | kfife | The filename even gets barfed out to the console when invoked, but I can't seem to get my grubby little dialplan hands on it. |
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| 23:30.29 | kfife | Is there a channel variable for the automixmon filename? e.g. features.conf *3, or can it be set? |
| 23:30.52 | kfife | The filename even gets barfed out to the console when the feature is invoked, but I can't seem to get my grubby little dialplan hands on it. |
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| 23:51.06 | mekarpeles | Is there a channel for #festival? |
| 23:51.20 | kfife | ? |
| 23:51.44 | kfife | core show application Festival |
| 23:51.51 | kfife | or is it festivus? |
| 23:53.37 | mekarpeles | festival text-to-speech |
| 23:54.47 | vo1pbx | cepstral for the win... festival sounds like Frankensteins Monster. |
| 23:54.51 | kunwon1 | kfife: you only need to post questions once, it sometimes takes people a while to respond. The variable MIXMONITOR_FILENAME will contain the filename used for recordings, or you can set it manually when you invoke the MixMonitor() application |
| 23:54.51 | kfife | That was a seinfeld joke |
| 23:55.02 | vo1pbx | kfife: a poor one |
| 23:55.02 | kunwon1 | for details on the latter, core show application mixmonitor should have what you need |
| 23:55.24 | *** part/#asterisk kharwell (~kharwell@user-24-214-15-130.knology.net) |
| 23:56.14 | kfife | kunwon1: This is automixmon. it appears to be different in scope. The MIXMONITOR_FILENAME doesn't have the filename that is spit to the console. |
| 23:56.29 | kfife | Presumably becaue it's a feature. |
| 23:58.30 | kunwon1 | kfife: interesting, i didn't know that - try to run DumpChan() on the channel once you start automixmon, that will show you -all- channel variables |
| 23:58.33 | kunwon1 | maybe it's undocumented |
| 23:58.37 | *** part/#asterisk mekarpeles (~mek@208.70.31.232) |