Re: [asterisk-users] Asterisk & Vitelity Invite issues

[Israel Gottlieb]

Could you please write the problem your having and not the reason to the
problem
Maybe the reason is something else

בתאריך 8 באוג׳ 2016 17:25,‏ "Tammy Firefly"  כתב:

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.

There is no firewall on this machine at all.

Thanks
--Tammy

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk & Vitelity Invite issues

[Matthew Jordan]

On Thu, Aug 11, 2016 at 9:04 AM, Tammy Firefly  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.

Matt

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

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk & Vitelity Invite issues

[Tammy Firefly]

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.





On 8/10/16 4:21 PM, Matt Fredrickson wrote:
> 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  
> wrote:
>>
>>
>> On 8/9/16 12:40 PM, Matt Fredrickson wrote:
>>> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly  
>>> 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.wis...@wiztech.biz
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk & Vitelity Invite issues

[Matt Fredrickson]

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  wrote:
>
>
> On 8/9/16 12:40 PM, Matt Fredrickson wrote:
>> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly  
>> 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.wis...@wiztech.biz
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk & Vitelity Invite issues

[Tammy Firefly]

On 8/9/16 12:40 PM, Matt Fredrickson wrote:
> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly  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.wis...@wiztech.biz

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk & Vitelity Invite issues

[Matt Fredrickson]

On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly  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?

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk & Vitelity Invite issues

[Tammy Firefly]

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.

There is no firewall on this machine at all.

Thanks
--Tammy

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users