[asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code

Thu Oct 9 08:25:48 CDT 2008

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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