Transfering Outbound call with Aastra 53i using TBPro Call Center Edition *** Update *** 3.0

I setup a grandsteam gxp200 and outbound calls were able to be transfered around to internal ext's. no problem with no additional configuration. Everything worked as expected. Now it appears to be isolated to aastra as HUD is able to tranfer outbound calls and the Grandstream will as well. The aastra phone was autoconfigured by tbpro. I am going to attempt to contact Aastra to see if they can assist me with this. If anyone has experienced this or has any ideas, I would love to hear them.

I will check that out. But you would think since it was autoconfigured by tbpro, it would by default be able transfer an outbound call. Very strange. Thanks for the reply. I will update when the issue is resolved.

I spoke the the Aastra Tech. He suggested putting the phone back to factory defaults. Did that. He had me manually configure the sip line 1 setting with username, password, etc. Did that. When the phone booted, I skipped the auto discovery. Then the phone works as it should. When I let trixbox hand out the config, we are right back to the beginning. Seems like whatever trixbox is handing my phones as far as the auto configuration is broken. The tranfer of a call that is placed from a phone to an outside line is the only thing that seems to be broken at this point. Maybe other things but they have not shown up. Another intersting thing is that by default setting, the transfer softkey is present and works. When the configuration is obtained from the server, all of the programming for the soft keys are blank. I can post some things relating to the astra tftp configs

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://s126699.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
#
# as of firmware 1.4 the tx gain setting should be zero
handset tx gain: 0
handset sidetone gain: 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|*[13]|9011X+#|85[05]5|[78]XXX|*[65]X|9[469]11|9[2-9]XXXXXXXXX|*[78][456]XXXX|91[2-9]XX[2-9]XXXXXX"
#
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

mac.cfg:

# The tftp_server must *not* be a hostname. Only IP addresses will work.
#tftp_server: s126699.trixbox.fonality.com
sip proxy ip: s126699.trixbox.fonality.com
sip registrar ip: s126699.trixbox.fonality.com

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 zone name: US-Central
#
displayName1: Scott Peterson
displayName2: 7001
#
sip line1 auth name: 00085D1B07AA
sip line1 password: 00085D1B07AA
sip line1 user name: 00085D1B07AA
sip line1 display name: 7001
sip line1 screen name: 7001
#
sip line2 auth name: 00085D1B07AA
sip line2 password: 00085D1B07AA
sip line2 user name: 00085D1B07AA
sip line2 display name: 7001
sip line2 screen name: 7001
#
sip line3 auth name: 00085D1B07AA
sip line3 password: 00085D1B07AA
sip line3 user name: 00085D1B07AA
sip line3 display name: 7001
sip line3 screen name: 7001
#
sip line4 auth name: 00085D1B07AA
sip line4 password: 00085D1B07AA
sip line4 user name: 00085D1B07AA
sip line4 display name: 7001
sip line4 screen name: 7001
#
sip line5 auth name: 00085D1B07AA
sip line5 password: 00085D1B07AA
sip line5 user name: 00085D1B07AA
sip line5 display name: 7001
sip line5 screen name: 7001
#
sip line6 auth name: 00085D1B07AA
sip line6 password: 00085D1B07AA
sip line6 user name: 00085D1B07AA
sip line6 display name: 7001
sip line6 screen name: 7001
#
sip line7 auth name: 00085D1B07AA
sip line7 password: 00085D1B07AA
sip line7 user name: 00085D1B07AA
sip line7 display name: 7001
sip line7 screen name: 7001
#
sip line8 auth name: 00085D1B07AA
sip line8 password: 00085D1B07AA
sip line8 user name: 00085D1B07AA
sip line8 display name: 7001
sip line8 screen name: 7001
#
sip line9 auth name: 00085D1B07AA
sip line9 password: 00085D1B07AA
sip line9 user name: 00085D1B07AA
sip line9 display name: 7001
sip line9 screen name: 7001

sip intercom type: 2
sip intercom prefix code: *75
topsoftkey1 type: speeddial
topsoftkey1 label: "Voicemail"
topsoftkey1 value: 8555
#
topsoftkey2 type: dnd
topsoftkey2 label: "DND"
topsoftkey2 value: dnd
#
topsoftkey3 type: callers
topsoftkey3 label: "Call List"
topsoftkey3 value: 6
#
topsoftkey4 type: icom
topsoftkey4 name: "Intercom"
topsoftkey4 label: Intercom
#
sip intercom type: 2
sip intercom prefix code: *75

Thanks so much. I will attempt to contact support. This has not worked since I put this server up and have been struggling with it. I wonder if TB will want to charge me for a support call even though the 53i is on their certified hardware list and the phone technically autoconfigures. After looking at the configs on the server, it appears that aastra.cfg file is not even for the 53i with regard to the programmable buttons. Do you think you could post out what the CE version of the aastra scripts look like? I am curious. I only have pro installed. Thanks again and I will post my findings.