Using chan_sip, we are able to register with an NEC switch.When I try to REGISTER with PJSIP, the authentication is being rejected.Traces show its using md5 authentication. The packets looks almost identical.The one area that I suspect is causing ..
Author : dcropp
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We have a customer using ConfBridges. Party A is connected, audio is fine. We originate a call to party B through an Avaya switch.It forwards the call to IVR. The two channels are added to the same ConfBridge.Using a wireshark capture, I can listen..
We are working with an Avaya switch.We send them a REFER.If the transfer is successful, everything is great.If it fails (busy), they send an INVITE in-dialog with a media attribute of inactive.After that, they send a 486 busy. The problem is Avaya basica..
We have a scenario where we have ConfBridges assigned to specific users. An administrator wants to listen to the owner of a ConfBridge calls and may want to whisper instructions to them.I thought ChanSpy would be the perfect solution for this.Dur..
Does anyone know if there is a way to disable the norefersub for PJSIP?It appears this is causing problems with a test were running with Cisco.A wireshark trace from a system where the transfer with Cisco works versus a trace with Asterisk/Cisco sh..
Running a test using asterisk 16.1.1 and two PCs with Firefox browsers.Im running the cmp2k demo.I place calls into the same asterisk and using AMI answer the calls and then add them into the same confbridge. Video mode is configured to follow_talker.Howev..
Using asterisk 16.1.1.Im setting up a test using the cmp2k (Cyber Mega Phone 2K Ultimate Dynamic Edition).I have noticed Chrome 72 had some issues with video streams.I just upgraded to Chrome 73 and see they still have some issues.If I have 2 calls..
Did something recently change for the chan_sip bindport setting?I know I had this working with the previous version of asterisk.Cant remember if it was an earlier 16.x version or 13.x I was running chan_sip (binding to port 5061) and PJSIP using ..
I am trying to run the CyberMegaPhone demo to see the WebRTC Video Conference demonstration from AstriDevCon 2017I have been able to make WebRTC work on this same box with SIPML5 demo but not the CMP2K.When I attempt to access the https://myip:8089/cm..
I had SIPML5 working with my Asterisk 16 last week.Not sure what I changed, but Im now receiving the following in asterisk whenever I try to login.Can anyone provide some guidance on what I should be looking at or how to diagnose the problem?[12/12 08:46:18.1..