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00:21.21 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); 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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01:24.11 | *** join/#asterisk lagzilla (lagzilla@2600:3c03::f03c:91ff:fe6e:5ed9) |
01:25.40 | lagzilla | Is it possible to connect a hosted extension to an asterisk server as a trunk? |
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02:03.17 | Samot | I mean, yes. You could. |
02:03.21 | Samot | But why? |
02:05.46 | lagzilla | cost mostly it's just to mess around at home, sip trunk is like $20 hosted ext is $6 |
02:07.21 | lagzilla | chan_sip.c: Call from '000b827b2c92' (my.host.tld:5060) to extension '100' rejected because extension not found in context 'from-my.host.tld' |
02:10.02 | Samot | Do you have a context called that? |
02:12.54 | lagzilla | nope googling how to add it lol |
02:14.12 | Samot | Wait, you're going to try and use a $6 voip account for your PBX trunk? |
02:14.24 | Samot | Why even bother with having a PBX? |
02:14.25 | lagzilla | yea lol |
02:15.03 | lagzilla | bordom, there is no practical reason |
02:16.22 | Samot | So that type of service is not the same as a SIP trunk... |
02:16.24 | Samot | So... |
02:16.51 | lagzilla | I know, I just wasn't sure if it's possible |
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07:54.44 | jkroon | has anybody here used myodbc 5.3? we keep getting connection failures ... which we cannot explain. |
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08:06.48 | drmessano | Which distro? |
08:07.01 | drmessano | Debian/Ubuntu? |
08:09.26 | drmessano | jkroon: ? |
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11:18.15 | pluesch0r | hi everybody. i've migrated to asterisk 13.13.1 and also changed our sip config over to pjsip, now i'm stuck by that dreaded custom-sip header problem. i've managed to set alert-info headers for specific PJSIP channels, but i'm not able to do that with queues. we're not using local agents or anything, can this be done somehow? |
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12:09.40 | ruied | Hello I'm using cli> originate to make a call test to a sip trunk but I need to change the callerid. It works fom a phone, but not seeing how to do it with originate. Now I have this working: originate sip/<external_num>@g9 extension 804@total_Odiaxere but can't setcallerid |
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13:04.29 | Samot | ruined: There is an actual setting called "callerid" for this for Originate. |
13:07.21 | [TK]D-Fender | not in CLI there isn't... |
13:07.30 | Samot | Hrm. |
13:07.37 | Samot | OK I generally don't do it from the CLI |
13:07.38 | [TK]D-Fender | AMI yes, Cll files yes, CLI no. |
13:07.48 | [TK]D-Fender | Call* |
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13:18.35 | ruied | [TK]D-Fender, ok. I will do it from AMI... thanks |
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14:55.12 | Kobaz | [TK]D-Fender: poke |
15:05.19 | [TK]D-Fender | yes? |
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17:20.41 | igcewieling | sometimes I feel sorry for our customers. The one I'm looking at had cable repairs done twice in 2 months, then a gas explosion took out their service. |
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17:29.34 | drmessano | igcewieling: Empathy must be painful for you |
17:29.41 | drmessano | Do you need to lie down? |
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