Hi I have searched the mailing lists and come across similar threads,
but no actual solution. I am trying to use a Cisco AS5300 as a
gateway for PSTNr. I have been able to configure it to take outbound
calls and send them to the PSTN just fine. Inbound calls however are
rejected by asterisk with "488 Not acceptable here" code.
here are the details:
AS5300:
IOS (tm) 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE
SOFTWARE (fc5)
Current configuration : 3939 bytes
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname K_AS5300_3
!
boot-start-marker
boot-end-marker
!
enable password ******
!
resource-pool disable
clock timezone EST -5
clock summer-time EDT recurring
!
no aaa new-model
ip subnet-zero
!
!
isdn switch-type primary-dms100
!
!
voice service voip
sip
bind all source-interface FastEthernet0
controller T1 0
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24
!
controller T1 1
framing esf
clock source line secondary 1
linecode b8zs
pri-group timeslots 1-24
!
controller T1 2
framing esf
linecode b8zs
pri-group timeslots 1-24
!
controller T1 3
framing esf
linecode b8zs
pri-group timeslots 1-24
!
!
!
interface Ethernet0
no ip address
shutdown
!
interface Serial0:23
no ip address
encapsulation hdlc
isdn switch-type primary-dms100
isdn incoming-voice modem 64
no cdp enable
!
interface Serial1:23
no ip address
encapsulation hdlc
isdn switch-type primary-dms100
isdn incoming-voice modem 64
no cdp enable
!
interface Serial2:23
no ip address
encapsulation hdlc
isdn switch-type primary-dms100
isdn incoming-voice modem 64
no cdp enable
!
interface Serial3:23
no ip address
encapsulation hdlc
isdn switch-type primary-dms100
isdn incoming-voice modem 64
no cdp enable
!
interface FastEthernet0
ip address 172.31.2.7 255.255.255.0
duplex auto
speed auto
!
ip classless
ip route 0.0.0.0 0.0.0.0 172.31.2.1
no ip http server
!
!
!
!
!
!
voice-port 0:D
!
voice-port 1:D
!
voice-port 2:D
!
voice-port 3:D
!
!
!
dial-peer voice 100 voip
application session
destination-pattern 678.......
signaling forward unconditional
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 101 voip
destination-pattern 770.......
progress_ind setup enable 3
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 102 voip
destination-pattern 404.......
progress_ind setup enable 3
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 103 voip
destination-pattern 470.......
progress_ind setup enable 3
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 200 pots
application session
incoming called-number .
destination-pattern 91..........
direct-inward-dial
port 0:D
prefix 1
!
dial-peer voice 201 pots
application session
incoming called-number .
destination-pattern 9..........
direct-inward-dial
port 0:D
!
dial-peer voice 202 pots
application session
incoming called-number .
destination-pattern 91..........
direct-inward-dial
port 1:D
prefix 1
!
dial-peer voice 203 pots
application session
incoming called-number .
destination-pattern 9..........
direct-inward-dial
port 1:D
!
dial-peer voice 204 pots
application session
incoming called-number .
destination-pattern 91..........
direct-inward-dial
dial-peer voice 204 pots
application session
incoming called-number .
destination-pattern 91..........
direct-inward-dial
port 2:D
prefix 1
!
dial-peer voice 205 pots
application session
incoming called-number .
destination-pattern 9..........
direct-inward-dial
port 2:D
!
dial-peer voice 206 pots
application session
incoming called-number .
destination-pattern 91..........
direct-inward-dial
port 3:D
prefix 1
!
dial-peer voice 207 pots
application session
incoming called-number .
destination-pattern 9..........
direct-inward-dial
port 3:D
!
sip-ua
retry invite 4
retry response 3
retry bye 2
retry cancel 2
sip-server ipv4:172.31.2.29
!
!
line con 0
line aux 0
line vty 0 4
password ****
login
!
ntp clock-period 17179848
ntp peer 192.43.244.18
end
Asterisk:
Asterisk 1.2.12.1 on a x86_64 running Linux
sip.conf:
[general]
context=default ; Default context for incoming calls
bindport=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
[as5300_1]
type=peer
host=172.31.2.7
permit=172.31.2.7/255.255.255.255
defaultip=172.31.2.7
disallow=all
allow=ulaw
allow=gsm
allow=alaw
nat=no
canreinvite=yes
dtmfmode=rfc2833
I have also included links to text files containing debug from both
asterisk and the as5300 for a successful outbound call as well as a
failed inbound call. Any help on gettign the inbound to work would be
great. Thanks in advance.
http://www.ketema.net/outbound_asterisk_debug.rtf
http://www.ketema.net/outbound_cisco_debug.rtf
http://www.ketema.net/inbound_debug_asterisk.rtf
http://www.ketema.net/inbound_debug_cisco.rtf
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