[asterisk-users] Asterisk & Vitelity Invite issues

Matthew Jordan mjordan at digium.com
Thu Aug 11 13:41:11 CDT 2016

On Thu, Aug 11, 2016 at 9:04 AM, Tammy Firefly <tammy-lists at wiztech.biz> wrote:
> my bad, both sides are generating re-invites.  Vitelity ignores any
> inbound invites to continue call flow.  to keep the call going our pbx
> has to deal with their re-invites otherwise the call terminates at 30
> minutes on the dot.  Our side is ignoring the inbound invites from
> vitelity and that causes the call to be torn down.

The 'directmedia' or 'canreinvite' settings only apply to Asterisk
generating a re-INVITE to initiate remote packet bridging. Setting
that to 'no' will only prevent Asterisk from initiating a re-INVITE to
perform said bridging; it won't apply to anything else. There's a
whole host of reasons why Asterisk would generate a re-INVITE. That
could be due to SIP session timers, or because a change occurred in
the party identification via a connected line update. Asterisk will
generate re-INVITEs when that happens, and there isn't a setting that
will prevent that from happening.

Asterisk should have no problem accepting and handling a re-INVITE
from a provider, so long as it is formed correctly.

If your provider can't accept a re-INVITE being sent to them, there's
something seriously wrong with that provider. This is pretty core
functionality in any SIP stack.


Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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