Hello,
I currently have this configuration:
PSTN <====> SIP/ALG Router <====> OpenSips 1.10 <====> IVR
OpenSips has a single IP on the private network.
I have configured opensips using top hiding in the dialog module and it
works fine for calls to ptsn and calls from pstn.
I have also configured opensips using B2BUA top hiding and it also works
fine for calls to ptsn and calls from pstn.
Now I want to test B2BUA REFER scenario (where calls from PSTN are
answered by IVR, then IVR does a REFER to another PSTN number).
When the IVR sends REFER the call is dropped after .6 seconds. The flow
that I've seen in the trace is below:
PSTN opensips IVR
invite+SDP (Call1) ----> |
<----- Trying (Call1) |
| invite+SDP (Call2)----->
| <----- OK+SDP (Call2)
<----- OK+SDP (Call1) |
Ack (Call1) ----> |
| ACK (Call2) ----->
<Ivr dialog take place here>
| <----- REFER (Call2)
<----- Invite (Call1) |
| Accepted (Call2) ----->
| BYE (Call2) ----->
Trying (Call1) ----> |
OK+SDP (Call1) ----> |
<----- Invite+SDP(Call3)|
| <----- OK (Call2)
Trying (Call3) -----> |
OK+SDP (Call1) -----> |
OK+SDP (Call1) -----> |
<0.6 seconds elapse here>
Bye (Call1) -----> |
<----- OK (Call1) |
<----- Cancel (Call3) |
OK (Call3) -----> |
Req Termd (Call3) ----->|
<----- Ack (Call3) |
It looks as though the PSTN times out waiting for an ACK after sending
OK+SDP(Call1) a couple times and then waiting .6 seconds.
The question is - what should the flow look like? According to this
post: http://lists.opensips.org/pipermail/users/2012-April/021352.html,
things appear to be working as expected up to the point where we receive
Trying (Call3). Should I be seeing the OK+SDP from call 3 next?
I'd like to troubleshoot further but I'm not sure where to look.
Thanks!
Tony
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