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

Alex Balashov abalashov at evaristesys.com
Thu Oct 9 08:36:57 CDT 2008 

This is due to an SDP mismatch of some sort, codec or otherwise.

Perhaps you have not set your Asterisk SIP peers to support RFC2833
DTMF?  Try dtmfmode=rfc2833.  Either that, or your Asterisk SIP peers
are not accepting the gateway's offer of G.711u.

Of course, I have seen interop bugs in Asterisk in the past where inbound
calls from Cisco ISDN gateways whose SDP payload advertises a different
preferred codec--but one that is still acceptable further down the
preference chain--is denied.  You may want to try to set both sides to
G.711u explicitly, i.e.

  disallow=all
  allow=ulaw

On the Asterisk side.  Also make sure dtmfmode is set.

On Thu, October 9, 2008 9:25 am, Ketema Harris wrote:

> 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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-- 
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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