VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
| View previous topic :: View next topic |
| Author |
Message |
ron at silverbackasp.com
Guest
|
 Posted: Fri Feb 22, 2008 5:50 am Post subject: [asterisk-users] canreinvite question |
 |
|
Hi All,
if i do this setup:
|---[ext 100]
|--[router/nat gw]--|
| |---[ext 101]
|
[asterisk]--[internet]---|
|
| |---[ext 200]
|--[router/nat gw]--|
|---[ext 201]
If i set, canreinvite=yes on all ext, assuming all ip phones have the
same codec, if 100 calls 101, or vice versa will rtp still go thru
asterisk? and same scenario for 200 to 202 or vice versa.
what if 100 call 200 or 201? or 200 calls 100 or 100? will rtp still go
thru asterisk?
thank you
regards,
Ron |
|
| Back to top |
|
 |
vincent.delporte at bi...
Guest
|
| Posted: Fri Feb 22, 2008 7:03 am Post subject: [asterisk-users] canreinvite question |
|
|
On Fri, 22 Feb 2008 18:50:16 +0800, Ron <ron at silverbackasp.com> wrote:
| Quote: | If i set, canreinvite=yes on all ext, assuming all ip phones have the
same codec, if 100 calls 101, or vice versa will rtp still go thru
asterisk? and same scenario for 200 to 202 or vice versa.
|
... and I'd like to add to this question: If the phones have the
option "Enable NAT", I expected them to be able to talk to each other
directly, but they didn't, and I had to set them to "canreinvite=no"
in sip.conf. Why is that? |
|
| Back to top |
|
|
|
|
|
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|
Search
