On Thu, 2008-04-10 at 11:13 -0500, Brent Davidson wrote: > You might also try "canreinvite=no" for both your phone and the sip > peer.
Yeah, there is definitely no re-inviting going on. Both Asterisk and the local handset are in a local network behind NAT with reference to the SIP server that requires INFO. > I think it's normal procedure for Asterisk to drop out of the > call path once the call is established between two peers. The > canreinvite directive forces asterisk to remain as an intermediary, and > it will probably do the transcoding that way. Indeed, this is my understanding as well, but I am definitely not getting a bridging of the sipphone and sip provider through a re-invite. The NAT would not facilitate it. > If I'm not mistaken this > is also useful for making calls between two system that have no common > codecs. Right. I need to use ekiga or one of the ATAs here that I know support rfc2833 so that I can eliminate this possible need to transcode inband to info/rfc2833 in order to narrow down the field. b.
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