[Asterisk-Users] this is the code that breaks outgoing calls on grandstream

Here is the diff from chan_sip.c 15 days ago and 16 days ago. 15 days ago is the point outgoing calls made via grandstream budgetone stopped working. Any help on why it breaks? Any possible fix? /tmp# diff asterisk/channels/chan_sip.c asterisk.works/channels/chan_sip.c 289d288 < int capab

Re: [Asterisk-Users] this is the code that breaks outgoing calls on grandstream

There would have to be a corresponding change in the SIP dialog or in the actual audio sent both ways. Can you provide some information on how it has changed? Mark On Fri, 7 Nov 2003 [EMAIL PROTECTED] wrote: > Here is the diff from chan_sip.c 15 days ago and 16 days ago. 15 days ago is the > p

Re: [Asterisk-Users] this is the code that breaks outgoing calls on grandstream

The broken code sends audio directly to the NAT address. For instance: When I place a call from the grandstream to * with the broken code * sees the grandstream. * according to tcpdump, sends a bunch of UDP (audio?) directly to the PRIVATE IP (192.168.0.100) of the grandstream even though it is