trixbox Pro: Long delay for connecting calls using Aastra phones

Long delay for connecting calls using Aastra phones

Hello All...I posted this in trixbox Pro help, but don't think that's a very active forum so I thought I'd try here instead,

Please see below, I'm hoping someone can point me in the right direction? Any help would be greatly appreciated.

Problem: When making a call on any of our Aastra phones, there is a long delay (10-15 seconds) from the time the call is dialed, until it actually connects and starts ringing.

Maybe helpful info:
* We have several Aastra phones (9133i and 480i) and they ALL exhibit this same issue.
* We also have a Polycom (501) that does NOT exhibit this issue.
* This issue has existed from the initial setup of Trixbox Pro.

Things I've tried:
* Various "dial plan" changes on the local phones themselves. Even to the point of using the default dial-plan that comes with the phone before it gets the configuration from the trixbox script (aastra.cfg)
* Tried deleting the .cfg files for one of the phones and let it reconfigure from a factory default reset This did not work either and resulted in having to dis-associate the extension, delete the phone, and reinstall it to get it to generate the new .cfg file.

Below is the dump from my aastra.cfg file that I understand these phones download their settings from (it's long, sorry, but I'm not sure what's going to be relevant and not):
----------------------------------
[root@trixbox117455 tftpboot]# more aastra.cfg
dhcp: 1
tftp server: 192.168.1.10
#time server disabled: 0
#time server1: tick.ucla.edu

# sidecar helper
sip xml notify event: 1
action uri xml sip notify: http://s117455.trixbox.fonality.com/phone/sidecar.cgi

sip digit timeout: 4
sip use basic codecs: 0
sip silence suppression: off

sip customized codec: payload=0;ptime=20;silsupp=off;payload=8;ptime=20;silsupp=off;payload=18;ptime=20;silsupp=off;

time server1: 164.67.62.194 # tick.ucla.edu
time server2: 216.218.192.202 # clock.fmt.he.net
time server3: 204.74.68.55 # ntp2.sf-bay.org
live dialpad: 1
#
sip allow auto answer: 1
sip intercom mute mic: 0
#
#
# old dp:
#sip dial plan: "0|*5X|*1|*3|*66|*7[4-5][1-8]XXX|*75[1-8]XXX|*86[1-8]XXX|911|9911|9411|9611|[1-8]XXX|900X|9[2-9]XXXXXXXXX|91[2-9]XXXXXXXXX|9011+.|X+.#"
#
sip dial plan: "0|911|900X|9011X+#|[78]XXX|*[78][456]XXXX|91480[2-9]XXXXXX|91623[2-9]XXXXXX|X[2-9]XX[2-9]XXXXXX|X1[2-9]XX[2-9]XXXXXX|XX+#"
#
prgkey1 type: speeddial
prgkey1 name: "VoiceMail"
prgkey1 value: 8555
#
prgkey2 type: dnd
prgkey2 name: "DND"
prgkey2 value: dnd
#
prgkey3 type: line
prgkey3 name: "Line 8"
prgkey3 value: 8
#
prgkey4 type: line
prgkey4 name: "Line 7"
prgkey4 value: 7
#
prgkey5 type: line
prgkey5 name: "Line 6"
prgkey5 value: 6
#
prgkey6 type: line
prgkey6 name: "Line 5"
prgkey6 value: 5
#
prgkey7 type: line
prgkey7 name: "Line 4"
prgkey7 value: 4
handset sidetone gain: 0
handset tx gain: 0
headset sidetone gain: 0
headset tx gain: 0
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