Hi all, I have a major issue with a codec renegotiation in an asterisk 1.4.33.1 setup, which leads me to ask a general question about asterisk 1.4.X codec negotiation: asterisk can support a re-negotiation of a codec "on the fly" through a re-Invite? If my SIP provider sends me a re-invite changing codec from g729 to g711, asterisk properly handle the matter? I see in the trace that asterisk responds 200 OK to the provider, but *does not send the re-invite to the UAC, and stops to send rtp to the UAC*.
In mantis/jira I see this issue: https://issues.asterisk.org/jira/browse/ASTERISK-17261?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel#issue-tabs which is similar, and Matthew Nicholson wrote this comment: "This is not the way fax passthrough works in asterisk. If you would like to do passthrough in this manner, set up the channel for ulaw or alaw from the start, or use T.38 passthrough. T.38 passthrough is the only fax passthrough configuration that we officially support". So asterisk does not support a codec change "on the fly" with a re-invite, unless it is a t38 re-invite? This behaviour is also present in new asterisk versions (eg asterisk 1.4.42)? Thanks in advance, Matteo
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