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| 00:13.37 | degenerate | Sorry if this is a dumb question, but if I dial a local number, and the owner of that number has configured it to call forward to another number that is long distance. Who pays the long distance? |
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| 01:19.55 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.5 (2017/12/22), Standard: 15.1.5 (2017/12/22); 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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| 01:38.15 | kitsune | I think most carriers would charge the forwarder and not the forwardee. |
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| 08:50.41 | Spajjie | Hi All, can somebody help with MixMonitor, I'm trying to run a simple command to pass details through to an API after the MixMonitor is finished... but I dont suspect anything is working.. |
| 08:51.25 | Spajjie | Here is my dialplan command; same=>n,Mixmonitor(${path}${EXTEN}_${UNIQUEID}.wav,i(monitorid),ls) |
| 08:51.37 | Spajjie | The LS command doesn't even run in this instance |
| 08:51.41 | Spajjie | Any ideas what could be wrong? |
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| 09:02.15 | tuxd00d | Is âlsâ located in the directory specified for âastagidirâ in asterisk.conf? |
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| 13:32.31 | polysics | hey! can I define 1000 peers all with the same password in PJSIP without copy-pasting each one of them? |
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| 13:53.17 | file | you can reduce the amount using templates, https://wiki.asterisk.org/wiki/display/AST/Using+Templates |
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| 16:23.39 | Kobaz | hmmm |
| 16:23.57 | Kobaz | when running MOH, does it use musicclass from that channel. or the bridgepeer channel |
| 16:27.33 | Kobaz | oh, i think my export is just wrong |
| 16:27.50 | rmudgett | There are several methods to select which MOH class is used. bridgepeer is not one of them. |
| 16:28.32 | Kobaz | k, good to know |
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| 20:11.06 | spenk | hi, Does the RFC 4028 fully supported or the UPDATE Method is still unsupported in the last Asterisk ? Sorry if this is the wrong place to this type of questions |
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| 20:19.44 | file | both chan_sip and chan_pjsip support SIP session timers |
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| 20:21.37 | spenk | hi file, i'm using an old version based on Elastix and Asterisk 1.8.x session timers are supported but my legacy pabx send an UPDATE message and not INVITE and Asterisk respond as "Method not implemented" |
| 20:22.07 | [TK]D-Fender | those versions are ancient and not supportes |
| 20:22.09 | [TK]D-Fender | d |
| 20:22.32 | velix | Hi there. Can I use Asterisk to dial a number by software and let my IP telephone ring/connect when the connection was successfully created? |
| 20:22.44 | spenk | ok ok i not want a support to these version :) only a confirm of what i see |
| 20:38.01 | [TK]D-Fender | velix, Yes. AMI Originate , Call Files, etc |
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| 20:52.40 | velix | [TK]D-Fender: Nice. I want to do some phone survey for my academic studies. Sorry for asking noob questions: Can I record calls, if the phoned person allows it? |
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| 20:55.01 | polysics_ | Hello everyone! What would the recommended way be to define a very large number of peers (10k) that have the same password? Itâs for a video conferencing application |
| 20:56.05 | robmal | o_O |
| 20:56.40 | polysics_ | robmal: I know :) |
| 20:57.12 | robmal | I would recommend to change the video conferencing application. |
| 20:57.19 | polysics_ | It might actually be better to look into realtime here. I was hoping for someone to know âthis one weird trickâ |
| 20:57.52 | robmal | I can has realtime peers, easy as insert into mysql. But the 'same password' part is mad. |
| 20:58.05 | polysics_ | In what regards? The only way to identify video streams is to have everyone log in as their own user |
| 20:58.32 | polysics_ | And there is no security involved here - it is literally just to identify people |
| 20:58.51 | robmal | As always |
| 20:59.35 | polysics_ | I wish there was a regex pattern way to define peer usernames :) |
| 21:00.01 | polysics_ | Like \d{5} |
| 21:00.07 | polysics_ | That would do it |
| 21:00.18 | robmal | As in _XXXXX ? |
| 21:00.56 | [TK]D-Fender | velix, yes |
| 21:01.23 | polysics_ | robmal: thatâs an extension pattern though, is it? |
| 21:01.39 | robmal | Indeed. |
| 21:01.50 | [TK]D-Fender | polysics_, Peers are peers |
| 21:01.58 | velix | [TK]D-Fender: Man... our university has a hardware VoIP device. It costs some thousand Euros to allow users to record calls. |
| 21:01.59 | [TK]D-Fender | you can already see it all in the sample config |
| 21:02.22 | velix | [TK]D-Fender: I'll setup Asterisk in the next days and play with it. |
| 21:02.33 | polysics_ | The ONLY thing that I can get to in the context where SDP for the SFU is created is the peer name |
| 21:03.09 | polysics_ | Ok, Iâll go read through the pjsip.conf samples |
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| 21:05.16 | polysics_ | If we just define 10k peers, will that make something catch fire? |
| 21:08.04 | robmal | I'm sure about chan_pjsip but chan_sip doesn't load the peers untill something tries to register with that username. |
| 21:09.54 | polysics_ | I think pjsip does the same thing |
| 21:10.01 | polysics_ | But we can try :) |
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| 21:29.14 | velix | Sorry for asking again. How many calls can I make via a 4-core raspberry? |
| 21:37.38 | [TK]D-Fender | a handful probably |
| 21:37.44 | [TK]D-Fender | depends what the calls are doing. |
| 21:37.50 | [TK]D-Fender | recording, transcoding,e tc all add load |
| 21:38.00 | [TK]D-Fender | if you're looking to do actualy work use an actual PC |
| 21:38.48 | [TK]D-Fender | If you're comparing to another solution that cost thousands don't start by asking how to use the PC equivalent of a wind-up toy as a PBX. |
| 21:39.08 | [TK]D-Fender | You don't need a huge server, but try to do better than a Pi |
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| 21:50.51 | spenk | thanks for all the work on Asterisk software. Great product. Byeee |
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