Archives : September-2016
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Consider the following scenario. A customers incoming call enters the system, and after some processing, the call is placed on a queue, where it will be picked up by an agent. Then, the agent makes an attended transfer (using asterisk internal transf..
Hellowhen setting a local forward (in this case to extension 23) on a SIP phone, I see the following on the Asterisk CLI :[Aug 31 14:59:34] — Got SIP response 302 Moved Temporarily back from 11.22.33.44:40670[Aug 31 14:59:34] — Now forwarding Local/myaccount184@CallFromQueue-000007f..
I have a device I wish to use that when activated will output RTPaudio 8kHz, 16-bit linear RTP audio.Can asterisk use that as a source and dial a phone so the user would hear that audio?Th..
Everyone,As many of you are already aware, we are rapidly approaching the time when the Asterisk 11 branch will go into what is known as security fix only mode.Up to this point, bug fixes have been included and merged into the 11 branch.For Aster..
The Asterisk Development Team has announced the release of Asterisk 13.11.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 13.11.0 resolves several issues reported by ..
,is there a way to use Wavecom gsm modem (type M1306b) with asterisk, and use it to send SMS ?I tried to used chan_dongle, but the modem cannot detected (it only detect huawei gsm modem usb).Thanks in advance…Regards,Ikka Jakarta -..
When I call my Asterisk server using a SIP client, the ICE server is used correctly if calling as a voice call, but the ICE server is not being used if calling as a video call.Please let me know what could be causing t..
I find that using hints with PJSIP on Asterisk 13 is very unreliable compared to regular SIP.I see many phones as unavailable when they are in fact available.Usually hints will work fine for a while after a phone registers but after a while it will rem..
I have an extension that looks like this:exten => 5555551111,1,Verbose(Door buzzer calling)same => n,Dial(SIP/user1&SIP/user2&SIP/user3)The idea is that any of the three users can answer the phone to let someone in.The problem is that if, say, us..
The Asterisk Development Team has announced the second beta of Asterisk 14.0.0. This beta is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 14.0.0-beta2 resolves several issues repor..