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After reviewing many other posts as well as wiki information on canreinvite and asterisk media path I am not clear on whether asterisk still manages sip signaling after a reinvite has been issued between a peer and a UA.
Here are the details;
UA <g.711u> Asterisk <g.711u> SIP long distance provider. The SIP LD provider uses a session border controller to ensure that all sip traffic originates from my asterisk IP address. The SIP LD provider will accept RTP streams from any source.
Due to an issue when sending faxes with * in the media stream, I want to remove asterisk from the media stream for specific UAs (faxes complete successfully without asterisk in the stream, tested by setting the UA to the asterisk IP address).
In theory, if canreinvite=yes, codecs match (g.711u) and there are no dial options that require asterisk to remain in the stream, the re-invite should be issued and the UA and the peer should be the endpoints of the RTP streams.
Questions;
Does it work? I am having trouble getting it to work that way. Is the sip signaling all handled by asterisk in this case? – required by my providers session border controller.
I guess what I am asking is can asterisk function as a SIP PROXY when configured correctly?
Any examples or limitations I might have missed?
Thank you!
Damon
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