Upgrade from freepbx 2.10 + Ast1.8 to freepbx 2.11 + Ast11 : various issues

FreePBX Installation / Upgrade
1

Hi all,

I am attempting an upgrade from FreePBX 2.10 with asterisk 1.8 to the latest distro.

I managed to migrate most of the data thanks to the back-up module.
A few things like the chan_mobile config was moved by hand.

Now I have a few issues, one is a beginner issue, but I could not solve it : There are no sound, even for a call placed between 2 extensions in the same subnet (so no Firewall issue).

Then I cannot solve the Google Voice ! This is a shame as the asterisk 11 integrates it “easily”.
Well I configured the Google Voice [Motif], and I appear connected. But when I place a call the call drops after a short while. In another post is the Log in case someone could help.
My google talk account does not have 10 digit number… So I filled the 10 digit field in freepbx with a land line number in the us.
I also looked on the firewall side but did not find anything wrong… I do not post the xmpp debug because its a little large.

Could those issues be linked to the fact that I imported the PBX config from a Asterisk 1.8 ?

2

[2013-04-24 15:32:06] VERBOSE[11807][C-00000006] pbx.c: – Executing [s@macro-dialout-trunk:30] Dial(“SIP/203-00000006”, “Motif/ggtalkvehiculosvipcom/[email protected],300,r”) in new stack
[2013-04-24 15:32:06] VERBOSE[11808][C-00000006] app_mixmonitor.c: == Begin MixMonitor Recording SIP/203-00000006
[2013-04-24 15:32:06] VERBOSE[11808][C-00000006] app_mixmonitor.c: == Begin MixMonitor Recording SIP/203-00000006
[2013-04-24 15:32:06] VERBOSE[11807][C-00000006] app_dial.c: – Called Motif/ggtalkvehiculosvipcom/[email protected]
[2013-04-24 15:32:06] VERBOSE[11807][C-00000006] app_dial.c: – Called Motif/ggtalkvehiculosvipcom/[email protected]
[2013-04-24 15:32:08] VERBOSE[11807][C-00000006] app_dial.c: – Motif/[email protected] is proceeding passing it to SIP/203-00000006
[2013-04-24 15:32:08] VERBOSE[11807][C-00000006] app_dial.c: – Motif/[email protected] is proceeding passing it to SIP/203-00000006
[2013-04-24 15:32:11] VERBOSE[11807][C-00000006] app_dial.c: – Motif/[email protected] answered SIP/203-00000006
[2013-04-24 15:32:11] VERBOSE[11807][C-00000006] app_dial.c: – Motif/[email protected] answered SIP/203-00000006
[2013-04-24 15:32:27] WARNING[11097] chan_sip.c: Retransmission timeout reached on transmission sfKy-EjPpO6KV-eCHNuVjhEPKunPrDeF for seqno 22256 (Critical Response) – See

3

Packet timed out after 15809ms with no response

4

is that normal that looking at the codec translation tables I have a value of 15000 in almost all cells ?
In my Asterisk 1.8 version I have much lower values…

5

checking the SIP Peers, i can see all the productions extensions, that are still connected to the old Asterisk server in 1.8.
But the testing server on which i am currently doing the migration has a different IP and none of the 30 IP Phones installed in the company has the IP of this test Asterisk server running the version 11.
How can it be that all those peers appear connected to the test server (they also appear in the peers in the Production server, and work normally on the production server) ?

6

i was a problem of duplicate MAC Adress on both prodcution and test servers.