[Asterisk-Users] cisco/asterisk interop issues?

hi, i have an issue that when making a call from a SIP phone going as follows: phone -- asterisk -- cisco(192.168.0.1) -- terminating voip platform(10.0.0.1) i get the cisco sending up an invite to the voip platform followed directly with a CANCEL message, as follows: Via: SIP/2.0/UDP

[Asterisk-Users] outbound sip calls on asterisk

hi, i would like all my calls originating from asterisk users bound for external to route to one destination, a session border controller. protocol used is sip. i have edited extensions_custom.conf with: exten = _.,1,dial(sip/[EMAIL PROTECTED]) would this be correct to send any calls from