Extention will not answer on new installs and outboud calls drop

Two seperate issues here one incoming and one outgoing

I get this error in the logs for the incoming call.

[2012-02-14 10:06:06] VERBOSE[22220] app_dial.c: – Connected line update to SIP/VOIP-trunk1-00000049 prevented.

I setup trunks of 4 different servers and used 3 different ATAs with same errors.
this exact equipment worked 6 months ago.

What would generate this error message for incoming calls.

Also with outgoing calls call connects fine voice at both ends but, if QUALIFY=YES call drops (times out)in 600ms, if set to QUALIFY=no it times out in 32000ms.
It appers there is an INVITE and a REPLY but the ACK fails to get through. NAT is working fine and registering in the sip debug or so it seems .

Was wondering if anyone could help with why the 2 problems occur and or a fix.
Thanks
V

We need a bit more info. What version of FreePBX? Distro or hand built? What type of trunking are you using.

It is distro FreePBX 2.8.1.4

It is sip trunking set for outgoing with the following:

host=xx.xx.xx.xx
type=friend
disallow=all
allow=ulaw&g729
dtmfmode=rfc2833
insecure=port,invite
sendrpid=no

ATA is dhcp. Server is connected to provider via static.

DID is user id and also the name of the sip extention.
thanks
Vince

Have you purchased g.729 licenses from Digium? Log excerpts would help also.

No purchase yet, figured it would pick up on ulaw and just skip the g729. Am I wrong?

I kept the logs to stubs. If more is needed I will be happy to submit. Just did want to dump a ton of junk on whoever.

Thanks
Vince

Logs from the incoming issue
[2012-02-14 20:22:01] VERBOSE[11729] pbx.c: – Executing [[email protected]:42] Dial(“SIP/VOIP-INNO-00000061”, “SIP/4405369486,20,trI”) in new stack
[2012-02-14 20:22:01] VERBOSE[11729] netsock2.c: == Using SIP RTP TOS bits 184
[2012-02-14 20:22:01] VERBOSE[11729] netsock2.c: == Using SIP RTP CoS mark 5
[2012-02-14 20:22:01] VERBOSE[11729] app_dial.c: – Called SIP/4405369486
[2012-02-14 20:22:01] VERBOSE[11729] app_dial.c: – Connected line update to SIP/VOIP-INNO-00000061 prevented.
[2012-02-14 20:22:21] VERBOSE[11729] app_dial.c: – Nobody picked up in 20000 ms

…And here is a stub from the outgoing

[2012-02-14 20:17:58] VERBOSE[11722] netsock2.c: == Using SIP RTP TOS bits 184
[2012-02-14 20:17:58] VERBOSE[11722] netsock2.c: == Using SIP RTP CoS mark 5
[2012-02-14 20:17:58] VERBOSE[11722] app_dial.c: – Called SIP/VOIP-INNO/4402286580
[2012-02-14 20:17:59] VERBOSE[11722] app_dial.c: – SIP/VOIP-INNO-00000060 is making progress passing it to SIP/4405369486-0000005f
[2012-02-14 20:18:07] VERBOSE[11722] app_dial.c: – SIP/VOIP-INNO-00000060 answered SIP/4405369486-0000005f
[2012-02-14 20:18:07] VERBOSE[11722] rtp_engine.c: – Locally bridging SIP/4405369486-0000005f and SIP/VOIP-INNO-00000060
[2012-02-14 20:18:39] WARNING[3242] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 102 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[2012-02-14 20:18:39] WARNING[3242] chan_sip.c: Hanging