Author : Jerry Geis

Home » Articles posted by Jerry Geis (paged 5)





I am running Asterisk 13.38.0 and using Conf Bridging.. one way audio. When the conf hangs up – sometimes all the endpoints do not hangup. I have rtptimeout` but that does not seem to help hang up the channel.Is there something else I should be sett..

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I am using asterisk 13.33.0 and POlycom phone with the latest firmware.The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted – it receives a call, after some time(couple minutes) it no longer receives a ri..

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Any using an OBI 202 (or equivalent 201, 212) is there an Auto Answer option in there and Im just not seeing – it or is it named something else ?Id like an incoming call to set two things:1) No ring2) Auto answerAnyone found a way to do those ?T..

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I see this device :Axis C8033 Audio Bridge Quick Specs:Communications Protocol: SIP. Ethernet Ports: 1x 10/100. PoE: 802.3af/at Type 1 Class 2. Additional Interfaces:Audio: one-way/two-way, mono. Audio Codecs: G.711, G.726, WAV, MP3. Edge Storage: micro..

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I have a call from a call file:Action: Originate Async: yes Channel: SIP/2012Codecs: ulaw,alaw,gsm Context: dialout Exten: callprogress Priority: 1Timeout: 20000Variable: SIPADDHEADER=Alert-Info: Ring AnswerActionID: 100014CallerID: Axis < 525 >The SIP/2..

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