Re: [asterisk-users] Asterisk SIP with no RTP audio port (was Internode weirdness)

2011-01-22 Thread Da Rock
On 01/22/11 22:04, Da Rock wrote: On 01/22/11 20:00, Da Rock wrote: On 01/21/11 20:28, Da Rock wrote: On 01/21/11 03:19, Tom Rymes wrote: On 01/19/2011 10:34 PM, Da Rock wrote: WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog

Re: [asterisk-users] Asterisk SIP with no RTP audio port (was Internode weirdness)

2011-01-22 Thread Da Rock
On 01/23/11 10:18, Da Rock wrote: On 01/22/11 22:04, Da Rock wrote: On 01/22/11 20:00, Da Rock wrote: On 01/21/11 20:28, Da Rock wrote: On 01/21/11 03:19, Tom Rymes wrote: On 01/19/2011 10:34 PM, Da Rock wrote: WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to