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

[John R Covert]

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.

Regards/john