[asterisk-dev] Provider requires Delay between OK and ReINVITE

[Paul Belanger]

On 12-08-22 10:52 AM, John R Covert wrote:
> Help, please.
>
> I have a client using a provider which has just installed a new
> SIP trunk into our Asterisk installation from a different platform.
> All works fine on their "Sonus" platform.  However, on their "Fusion"
> platform, they drop a call when they receive a ReINVITE.
>
> 09:48:58.495468 IP PROVIDER > ASTERISK: SIP, length: 1001
> INVITE sip:exten at asterisk:5060 SIP/2.0
>
> 09:48:58.501365 IP ASTERISK > PROVIDER: SIP, length: 503
> SIP/2.0 100 Trying
>
> 09:48:58.886713 IP ASTERISK > PROVIDER: SIP, length: 519
> SIP/2.0 180 Ringing
>
> 09:49:12.794858 IP ASTERISK > PROVIDER: SIP, length: 815
> SIP/2.0 200 OK
>
> 09:49:12.795835 IP PROVIDER > ASTERISK: SIP, length: 397
> ACK sip:exten at asterisk:5060 SIP/2.0
>
> 09:49:12.801258 IP ASTERISK > PROVIDER: SIP, length: 835
> INVITE sip:2167129661 at 208.93.226.215:5060 SIP/2.0
>
> 09:49:12.801595 IP PROVIDER > ASTERISK: SIP, length: 336
> SIP/2.0 100 Trying
>
> 09:49:12.845627 IP PROVIDER > ASTERISK: SIP, length: 508
> SIP/2.0 491 Request Pending
>
> 09:49:12.848064 IP PROVIDER > ASTERISK: SIP, length: 614
> BYE sip:exten at asterisk:5060 SIP/2.0
>
> They have asked me to provide a delay between the "OK" sent
> by Asterisk and the ReINVITE.
>
> Can this be done in the existing code?  If I were to develop a
> change to SIP to allow this to be configured on a per-peer
> basis or some other way (please suggest) would it be accepted
> into the code base.
>
> I have been insisting that they either fix what seems to me to
> be a bug in their "Fusion" platform or put the new trunk back
> into the working "Sonus" platform, but they are pushing back
> on me.  Seems stupid, since we're paying them for some 10,000
> minutes per month of traffic and are offering to increase that
> five-fold.  But that's barely relevant to the technical
> discussion.
>
In the patch capture you provided, there is no REINVITE.  Are you 
talking about the INVITE message from asterisk to provider?

If I was to guess looking at this, call come into your asterisk box from 
ITSP, then you answer and send that call back out to another (same?) 
ITSP?  Is that correct?

Best you could do is:

[incoming]
exten => s,1,NoOp()
same => n,Answer()
same => n,Wait(5) ; Or whatever you want the time to be
same => n,Dial(SIP/ITSP/2167129661)

Also, if this is the case, this post belong on the asterisk-users 
mailing list.

-- 
Paul Belanger | PolyBeacon, Inc.
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