Re: [asterisk-users] Getting Asterisk out of the RTP media path

Russell Bryant wrote: On Aug 11, 2008, at 12:04 PM, SIP wrote: SIP wrote: When calling from our SIP proxy through Asterisk to the PSTN provider, we support reINVITES which tend to work seamlessly. However, when creating a call file which essentially connects a call from the

Re: [asterisk-users] Getting Asterisk out of the RTP media path

SIP wrote: When calling from our SIP proxy through Asterisk to the PSTN provider, we support reINVITES which tend to work seamlessly. However, when creating a call file which essentially connects a call from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP media path. I

Re: [asterisk-users] Getting Asterisk out of the RTP media path

On Aug 11, 2008, at 12:04 PM, SIP wrote: SIP wrote: When calling from our SIP proxy through Asterisk to the PSTN provider, we support reINVITES which tend to work seamlessly. However, when creating a call file which essentially connects a call from the SIP proxy to the SIP proxy,

[asterisk-users] Getting Asterisk out of the RTP media path

When calling from our SIP proxy through Asterisk to the PSTN provider, we support reINVITES which tend to work seamlessly. However, when creating a call file which essentially connects a call from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP media path. I understand that