Aastra phones not ringing

Hi All,

Spent most of the last two days googling and forum surfing to find an answer to this! Pretty sure I am missing something straight infront of me :smiley:

I have setup an Asterisk test rig with 5 SIP phones, 3 from a company called Safecom and 2 from Aastra (a 480i and a 9133i).

The Safecom phones are all working fine, I can ring between them quite happily. With the Aastra phones though I can only make calls from them. If I ring either of the Aastra phones I get no ringing sound on the line and the Aastra phone doesn’t ring. Within the Asterisk console whenever I ring an Aasta phone I don’t get a SIP/xxx is ringing event.

Safecom -> Safecom

  • Called 101
  • SIP/101 is ringing
  • Nobody picked up in 10000ms

Safecom -> Aastra

  • Called 103
  • Nobody picked up in 10000ms

I have followed some other config posts for the aastra phones on this forumn thinking it may be a phone config issue but to no avail. Any pointers would be most appreciated.

=== Extensions.conf

;###############################################
; Extensions configuration file
; /etc/asterisk/extensions.conf
; Created 02/08/06
;###############################################

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]

[internal]

; — Voicemail

exten=>399,1,VoiceMailMain()

; — SIP Phones

exten=>100,1,Dial(SIP/100|10)
exten=>100,2,Voicemail(u100)

exten=>101,1,Dial(SIP/101|10)
exten=>101,2,Voicemail(u101)

exten=>102,1,Dial(SIP/102|10)
exten=>102,2,Voicemail(u102)

exten=>103,1,Dial(SIP/103|10)
exten=>103,2,Voicemail(u103)

exten=>104,1,Dial(SIP/104|10)
exten=>104,2,Voicemail(u104)

=== SIP.conf

;########################################
; Test SIP server
; /etc/asterisk/sip.conf
;;########################################

[general]
context=default
realm=home.int
bindaddr=0.0.0.0
tos=lowdelay
maxexpiry=3600
defaultexpiry=120
musicclass=default
language=en

[100]
type=friend
context=internal
username=100
secret=testone
callerid=100
fromuser=100
host=dynamic
nat=no
mailbox=100
canreinvite=no
callgroup=2
pickupgroup=2

[101]
type=friend
context=internal
username=101
secret=testtwo
callerid=101
fromuser=101
host=dynamic
nat=no
mailbox=101
canreinvite=no
callgroup=2
pickupgroup=2

[102]
type=friend
context=internal
username=102
secret=testthree
callerid=102
fromuser=102
host=dynamic
nat=no
mailbox=102
canreinvite=no
callgroup=2
pickupgroup=2

[103]
type=friend
context=internal
username=103
secret=testfour
callerid=103
fromuser=103
host=dynamic
nat=no
mailbox=103
canreinvite=no
callgroup=2
pickupgroup=2

[104]
type=friend
context=internal
username=104
secret=testfive
callerid=104
fromuser=104
host=dynamic
nat=no
mailbox=104
canreinvite=no
callgroup=2
pickupgroup=2

Many thanks

do the aastra’s register? post your aastra config files…
also do a sip show peers and post that…

Pretty sure the phones are registered as I can ring the safecoms from the aastras.

Ext 100 to 102 are safecom phones
Ext 103 to 104 are aastra phones

=== sip show peers

Name/Username Host Dyn NAT ACL Port Status
104/104 192.168.1.31 D 5060 Unmonitored
103/103 192.168.1.30 D 5060 Unmonitored
102 192.168.1.34 D 5060 Unmonitored
101 192.168.1.33 D 5060 Unmonitored
100 192.168.1.32 D 5060 Unmonitored

=== aastra.cfg

Aastra GLOBAL phone configuration file

aastra.cfg

#================
#Network Settings
#================
dhcp:0

#====================
#Time Server Settings
#====================

time server disabled: 1
time server1:
time server2:
time server3:

#============================

Additional Network Settings

#============================

sip rtp port: 8000

#======================

Phone Update settings

#======================

download protocol: TFTP
tftp server: 192.168.1.85

#===================

Dial Plan Settings

#===================

#=====================

General SIP Settings

#=====================

sip session timer: 30
sip transport protocol: 0
sip use basic codecs: 1

#==============

Ring Settings

#==============
ring tone: 0
tone set: UK

=== mac.cfg

Aastra phone configuration file

Ext : 103

Edited :

Editor :

#=================

Network Settings

#=================

ip:192.168.1.30
subnet mask:255.255.255.0
default gateway:192.168.1.1
dns1:10.30.1.10

#==========================

Phone Global SIP Settings

#==========================

#— Change this section

sip user name:103
sip auth name:103
sip password:testfour
sip screen name:103
sip display name:103

#— End of change section

sip vmail:399
sip mode:0
sip proxy ip:uranus.home.int
sip proxy port:5060
sip registrar ip:uranus.home.int
sip registrar port:5060
sip registration period:3600

Many thanks

Looks okay to me… do a sip debug and post it

at cli do sip debug peer peername (ie 104)
then call that sip phone and post wht comes up on cli