Quote:
Originally Posted by emoci
A few things to try: ...
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Hmm...my earlier reply never got posted, so hoping this makes it. Thanks Emoci for the tests and additional methods.
On echo tests, sip:1234@loligo.com and sip:*850301@sipbroker.com work in all contexts. The one at sip:*010*600@sipbroker.com though only works when the appropriate SIP headers contain a public IP address. So to ensure maximum compatibility, one's client should be able to obtain his public IP (probably via STUN) and announce with it, and/or one should manage outgoing calls through a registered account (such as MySIPSwitch.com). The call-me echo test did not work, probably because I am not on a Voxalot account.
I have the PSTN gateway playing now. I finally determined that the call transfer itself was not the problem, but was because the inbound RTP stream beat my new outgoing RTP stream, thus grabbing the router's IP tables connection that I needed. The solution: turn on the SPI firewall in DD-WRT, so that any RTP stream that beats mine is dropped before it enters. (The firewall is on by default, but I thought I knew better and had it off.) Forwarding RTP ports is an alternative if one really has to have the firewall off.
The Toronto PSTN number works very well, but is time limited to just over 3 minutes. I also tested TPad (the number works, but account is no good) and Bezecom (works, but not yet sure of maximum call duration).
Thanks again, and hope this helps others!