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| 10:10.08 | wyoung | hihi |
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| 13:48.19 | _boot | has anyone gotten a grandstream ht701 to work with srtp? tls signalling works fine but i get a 401 unauthorized (my old friend) with encryption=yes, though in the config i've set the ht701 to use and force srtp, and RTP/SAVP is appearing in the sdp messages |
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| 15:49.04 | _boot | Odd, not sure if its a quirk of the HT701 but with insecure=invite, that peer can make srtp calls... Does this mean that someone knowing the peer's username could make a call through asterisk? or does it still require the host/port to match as registered by the peer? |
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| 15:54.49 | Ahmadi | Hi |
| 15:55.53 | Ahmadi | When i dial a call, if the destination number is busy, i have no busy message and the call wait until the destination number be free. how can i solve that problem? |
| 15:56.25 | Ahmadi | i need to detect that the destination of call is busy or not |
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| 18:02.14 | denysonique | Ahmadi: DIALSTATUS |
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| 18:10.02 | [TK]D-Fender | Not what he's looking for... |
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| 19:59.55 | denysonique | How can I originate a call with caller id set? |
| 20:00.59 | denysonique | ie Originate(sip/foo/07910001000,exten,context,1,1) |
| 20:04.18 | mmlj4 | set it? |
| 20:04.26 | [TK]D-Fender | use AMI or Call Files |
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