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| 10:50.05 | anexit | Is it possible to use G729 codec just for internal calls and use G711U for external? |
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| 14:12.33 | emsjessec | anyone know what the 1st interval and sub interval are on this table? https://www.flowroute.com/outbound-rates/?search=united+states#table |
| 14:13.32 | file | means that the minimum billable is 6 seconds, and then subsequently it is billed in 6 second increments |
| 14:15.46 | Samot | 6 is standard and good. |
| 14:16.29 | Samot | 6x6 that is. |
| 14:16.36 | Samot | Some do 30x6. |
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| 14:45.31 | bananapie | Hi, I am trying out asterisk and see that if I set dtmf=inband and send the caller into a context where they can dial numbers, it works perfectly. However, when I dial a sip endpoint that expects rfc2833(4755) and that endpoint is configured appropriately, asterisk does not detect the inband DTMFs and does not send a RFC2833 event out on the sip endpoint. How do I get asterisk to detect DTMF and convert to rfc2833 after a |
| 14:45.31 | bananapie | call is connected to sip end point. Thanks |
| 14:48.59 | Samot | The dtmfmode is for when that endpoint is sending DTMF. |
| 14:50.51 | bananapie | So I can't get asterisk to do dtmf conversion ( from inband to rfc2833 ) on a bridged channel ? |
| 14:54.34 | Samot | You're calling from Phone -> Asterisk -> PSTN? |
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| 15:15.05 | Syco | Hello people... I have a question about Dial and pjsip: with the old sip it was possible to dial SIP/username:secret@host!tonumber to use authentication without a peer in sip.conf. Can I achieve the same with pjsip?? |
| 15:16.29 | file | no. |
| 15:16.55 | file | that's not something anyone has implemented or created as of this time |
| 15:18.26 | Syco | Thanks |
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| 17:41.46 | as3ty | In a multiple trunks outbound route, how does one set a timeout so a call will jump to the next trunk if the connection time exceed x seconds? |
| 17:42.30 | as3ty | I should mention I am using FreePBX. |
| 17:46.14 | file | if you are using the GUI then you would want the #freepbx channel |
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| 19:35.36 | tpdmoore | hello - i'm using res_config_curl and dbsep.cgi, does this support voicemail messages, or is that still the exception? |
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