This was asked many years ago but I thought I would check to see if things have changed.Is it possible to take over a call in progress – using a replacement Asterisk server? In other words, if 2 user agents are connected through an Asterisk PBX, ..
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Is there a way to force a SIP client to re-register using a SIP command (or an AMI command)? If not, is there some other standard way to do so – or would I have to post/get to a web GUI of the phone (unique to each model) to force a reset, ..
I noticed that Asterisk 14 has changed the output format for some commands(eg: Output: ).However, the AMI reports version 2.8.0 which is the same as Asterisk 13 Is that considered a bug?Since the AMI output format has changed, shouldnt the AMI vers..
When I issue a sip show peers command the left most column is titledname/username.Some lines show one item in the column like 123, others show bob/123.Can someone explain the difference? (What does does name vs username mean) And why does sip show us..
We are working with an ISP that needs Asterisk to place a FQDN name in the SIP FROM and INVITE fields – where Asterisk is currently using an IPaddress.A SIP trace shows the following from my Asterisk box: INVITE sip:62351155@1.1.1.1 SIP/2.0Via: SIP/2.0/..
I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring at the handset.I captured the SIP traffic and see that my M65 is replying with an 488 not acceptable here.From what I read this is usually codec related but b..