[asterisk-users] Asterisk & Vitelity Invite issues

Matt Fredrickson creslin at digium.com
Wed Aug 10 17:21:21 CDT 2016


Wait a second, I thought in your original email that you said that
Asterisk was generating reinvites.  It sounds now like you're saying
that the remote side is initiating reinvites instead.

My understanding is that the canreinvite/directmedia option only
influences Asterisk's behavior with regards to generating reinivites.
If it receives a reinvite, I don't think these options will do
anything about that.  In fact, I'd guess that not properly responding
to a received reinvite is going to potentially break things from the
SIP perspective.

Matthew Fredrickson


On Wed, Aug 10, 2016 at 4:53 PM, Tammy Firefly <tammy-lists at wiztech.biz> wrote:
>
>
> On 8/9/16 12:40 PM, Matt Fredrickson wrote:
>> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-lists at wiztech.biz> wrote:
>>> Hi All,
>>>
>>> We have asterisk 11.23 running sip to vitelity and from there IAX trunks
>>> split off to where they need to go.  We are having a problem getting
>>> chan_sip to quit ignoring re-invites from Vitelity.  Our side ends up
>>> sending a reinvite which their side & they do not support us sending a
>>> reinvite.  Ive tried:
>>>
>>> canreinvite=no which was supposedly replaced by:
>>>
>>> directmedia=no
>>>
>>> Can anyone shed any light on this matter?  I'd love to get this fixed.
>>>
>>
>> Those options *should* influence chan_sip's reinvite behavior - at
>> least they have from my experiences working with chan_sip.  Do you
>> know what is triggering the reinvite in the first place, or does it
>> look like a normal media reinvite?
>>
>
>
> every 15 minutes vitelity sends a re-invite to keep the call going.  I
> have a packet capture from it if you'd like it feel free to email me off
> list @ tamara.wisdom at wiztech.biz
>
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-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA



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