Title says it all – for the time being I am stuck deploying Asterisk in ESXi . We are also looking at Proxmox for our next round of servers.. Everything works fine except conference calls – very stuttery , have tried a few different codecs.I assume t..
Author : Kevin Long
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My company has invested heavily in Counterpath’s Stretto provisioning platform for Mobile and Desktop VoIP clients .At this time their system allows 2 devices (for example iPhone + desktop computer) using the same software license per user , wh..
I am having trouble with RTP and NAT :Below is a SIP SDP invite from a remote endpoint which is trying to call extension 420 which is the ECHO application .As you can see, the public IP is where the request comes in from,but the SDP contains the priva..
Greetings.I am using the PJSIP driver with TLS transport, and my endpoints are SIP mobile apps operating in environments that I do not control.I would like Asterisk to default to sending INVITES and all other SIP signals to endpoints via the exist..
I use TLS transport for all my endpoints on my production system (Asterisk 11) .I need to debug some NAT traversal issues, and would like to use the ‘sngrep’ tool which shows SIP messages from a packet capture.Per the developer of ‘sngrepâ€..
My asterisk systems sit behind a Meraki mx80 firewall at a data center.I use static public IPs on the firewall and port forward5060,5061, and 10,000-20,000 so the clients can connect. Per Meraki support: Our MX security appliances do not support ..