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| 06:55.11 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.3.0, 16.17.0 (2021/03/25) Final Bugfix: 13.38.2, 17.9.3 (2021/03/04); 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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| 09:15.02 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.3.0, 16.17.0 (2021/03/25) Final Bugfix: 13.38.2, 17.9.3 (2021/03/04); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 09:17.15 | *** join/#asterisk zamba (~marius@89.162.88.18) |
| 09:17.22 | zamba | is asterisk still alive? |
| 09:25.53 | LiuYan | it's alive. why came out this question? |
| 09:29.04 | post-factum | mine one is alive too |
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| 14:02.09 | seanbright | is there a way to set a variable on another channel from dialplan? |
| 14:02.29 | Samot | New channel or existing channel? |
| 14:02.37 | seanbright | existing |
| 14:02.54 | Samot | Because you can do IMPORT() to pull vars from another channel |
| 14:03.11 | Samot | Oh wait, sorry. Set. |
| 14:03.15 | Samot | SetVar? |
| 14:03.29 | seanbright | Set? |
| 14:03.29 | Samot | No, now I'm being dumb. |
| 14:05.26 | Samot | And Set() will do the current channel. |
| 14:05.39 | igcewieling | SHARED() might work, depending on what you are trying to do. |
| 14:06.22 | *** join/#asterisk kharwell (uid358942@gateway/web/irccloud.com/x-stuqoacdyeacssgv) |
| 14:06.23 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:06.42 | Samot | Yeah, that's what I was trying to think of. |
| 14:07.39 | seanbright | interesting. i can't really "touch" the other channel at all, so i don't think this will work. but it's good to know about. |
| 14:08.04 | seanbright | basically, channel B is sitting in MoH |
| 14:08.19 | seanbright | channel A comes in to dialplan, does some stuff, and then calls Bridge(B) |
| 14:08.42 | igcewieling | I suppose you could use global variables with some sort of channel-specific identifier if you wanted a really ugly hack. |
| 14:08.57 | seanbright | i was thinking that too, but i would have no way to grab it from channel B |
| 14:09.19 | seanbright | channel B is being "pulled in" to a bridge by virtue of channel A calling Bridge() |
| 14:09.29 | Samot | Well if you knew channel B you could IMPORT |
| 14:10.03 | seanbright | IMPORT is pull, i need push |
| 14:10.07 | seanbright | unless i am misunderstanding |
| 14:10.29 | Samot | You want channel A to get something from Channel B? |
| 14:10.38 | seanbright | no |
| 14:10.43 | Samot | Vice versa? |
| 14:10.45 | seanbright | yes |
| 14:11.05 | seanbright | channel B is just sitting in MoH until yanked in to the bridge by channel A |
| 14:12.17 | seanbright | hmm, i could ChannelRedirect |
| 14:12.44 | seanbright | so maybe i would have an extension that would set the variable and then return the channel to MoH |
| 14:12.47 | seanbright | so i could do: |
| 14:12.59 | seanbright | ChannelRedirect(B, doTheThing) |
| 14:13.02 | seanbright | Bridge(B) |
| 14:13.13 | seanbright | i'm sure there would be no subtle timing problems with that |
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| 14:35.47 | *** mode/#asterisk [+o bford] by ChanServ |
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| 15:26.16 | Samot | seanbright: Did you catch my thank you the other week? |
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| 16:12.14 | seanbright | Samot: i did not |
| 16:13.05 | seanbright | Samot: just saw it - no problem |
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| 16:40.28 | Rem|1 | I'm having an issue with voicemail realtime whenever i restart asterisk the voicemail messages to the pjsip endpoints resets to 0, if someone leaves a new message to voicemail the number of messages are accurate. Is there something I am missing with a config somewhere? |
| 16:42.49 | *** join/#asterisk MLC (~MLC@63.249.40.11) |
| 16:42.56 | MLC | PJSIP - when I update the keys pointed to by cert_file= and priv_key_file= (transport) are those effectively immediately, or do I need to do something to reload? |
| 16:43.50 | file | reloading of certificates is not supported without allow_reload currently |
| 16:43.56 | file | restart is required otherwise |
| 16:44.35 | MLC | bummer |
| 16:50.27 | micdud | trying to make a direct dial call internaly to (an ata that is registered to an external provider, stun and all) ; ata rings but routing in asterisks gets assigned to external ip of the ata , only field with external ip from ata is Contact: ,, any ideas ? |
| 16:50.42 | *** part/#asterisk MLC (~MLC@63.249.40.11) |
| 16:51.53 | igcewieling | disable nat settings on the ata endpoint config. |
| 16:56.41 | micdud | set to nat=no ; but not sure if it picks it up as endpoint without it registering to asterisk |
| 16:57.11 | micdud | as it is already registered to another provider |
| 16:57.29 | Samot | So why are you trying to do this? |
| 16:58.14 | igcewieling | nat=no was deprecated like 10 years ago. |
| 16:58.35 | micdud | its only internal ata in the house , and want to route calls to it from another provider |
| 16:58.53 | igcewieling | micdud: I don't see Asterisk involved in that description |
| 16:59.25 | micdud | asterisk is registered to provider 1, ata to provider 2 , want to route some calls from asterisk to ata |
| 16:59.55 | Samot | The use the second line of the ATA to connect to Asterisk. |
| 17:00.04 | micdud | one line phone |
| 17:00.20 | Samot | Then you do proper call forwarding at the carrier/provider level. |
| 17:00.32 | micdud | without going outside |
| 17:00.39 | Samot | Not going to happen. |
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| 17:56.57 | *** join/#asterisk dacod (~dacod@191.243.8.176) |
| 18:02.19 | dacod | Hello everybody |
| 18:02.19 | dacod | I'm having trouble accessing wiki.asterisk.org, could someone confirm that it is accessible |
| 18:04.54 | file | it's not - http://lists.digium.com/pipermail/asterisk-users/2021-April/295778.html |
| 18:06.59 | dacod | thks |
| 18:10.32 | micdud | Samot even if i disable stun feature on the ata ? softphone clients have no problems connecting to that provider without stun from behind nat , so maybe ata will too. And then it will not represent itself as being behind nat . |
| 18:11.02 | Samot | Not true at all. It is behind NAT. |
| 18:11.18 | Samot | The STUN server is there to ensure the NAT is handled properly. |
| 18:11.57 | micdud | softphone seem to work fine without , to that provider though . |
| 18:12.06 | Samot | And you can forward calls there? |
| 18:12.26 | micdud | to softphones ? |
| 18:12.29 | Samot | What you are trying to do works if you are using a softphone and not the ATA. Just doesn't work with the ATA? |
| 18:13.15 | micdud | have not tried direct call to softphone , trying to localy to ata |
| 18:13.36 | Samot | There you go. Test that. |
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| 19:04.17 | zamba | do you guys recommend going the freepbx route to deploy asterisk? |
| 19:04.38 | file | like everything, depends on needs |
| 19:05.33 | zamba | well.. just want to get up and running.. basic sip trunk in.. some extensions.. need IVR.. |
| 19:06.03 | zamba | but i want to run it on an existing server.. so i guess i have to install it as a virtual machine to use the iso |
| 19:06.16 | igcewieling | you want a PBX, there is nothing simple about that. |
| 19:06.51 | zamba | igcewieling: well, i'm a bit used to PBX already.. so i know the basic concepts.. |
| 19:08.39 | igcewieling | If you use Asterisk, you'll need to understand Asterisk. If you use FreePBX you'll need to understand FreePBX and Asterisk, but you get a GUI so it makes you feel better about it. |
| 19:09.07 | zamba | the "Download Full ISO Now" link on asterisk.org seems broken |
| 19:09.35 | file | it does a popup thingy |
| 19:09.42 | file | same as https://www.freepbx.org |
| 19:10.05 | zamba | yeah, but nothing's downloaded |
| 19:10.21 | file | does the FreePBX one work? |
| 19:10.44 | zamba | i managed to get it from the download links section |
| 19:16.56 | igcewieling | turn on scripts. they LOVE scripts. |
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| 19:51.15 | Samot | So for queues, is it still only chan_sip can report 'in use' or is the sample docs just not updated? |
| 19:51.58 | file | others should too |
| 19:52.17 | Samot | Like chan_pjsip should be cool? |
| 19:52.19 | file | yes |
| 19:52.24 | Samot | Sweet. |
| 19:52.44 | Samot | I can fully Old Yeller chan_sip. |
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| 20:18.28 | edeneye | Hi, can someone help me about some issues in my config ? |
| 20:23.35 | seanbright | ~ask |
| 20:23.35 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:29.55 | edeneye | Ok but actually I don't see what precise point to expose to you. I am trying to config asterisk for Twilio via Zoiper 5 but nothing seems to work. I don't understand I am on it since like 5h and still don't catch it, its starting to make me crazy, really. Does someone have a model that work for this or anything I can base my config on ? |
| 20:30.43 | edeneye | I read some documentations for sip.conf but same, nothing seems to work |
| 20:49.09 | edeneye | ? |
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| 21:28.13 | Edeneye | Does anyone have a clue about what I need ? |
| 21:37.48 | Samot | You want to configure Asterisk to work with Twilio and Zoiper? |
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| 22:05.26 | Edeneye | Samot yes |
| 22:06.34 | Edeneye | I was being optimistic about the time I passed on it today. It's been 9h that I search everything I can search without results |
| 22:06.55 | Edeneye | Can you help ? |
| 22:08.54 | Samot | Dude. |
| 22:08.57 | Samot | Uncool. |
| 22:09.15 | Samot | You did not need to send me messages here and just randomly start PMing me. |
| 22:10.55 | Edeneye | Sorry I wasnt sure You'll read here. First time I use freenode |
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