[Asterisk-Users] SIP signaling and canreinvite=yes

[Damon Estep]

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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