[asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code
Ketema Harris
ketema at ketema.net
Thu Oct 9 09:26:08 CDT 2008
dtmf mode was set in the sip.conf
dtmfmode=rfc2833
I will remove the other codecs from sip.conf and see what effect it
has. Do you see any other potential issues in the configs?
thanks
On Oct 9, 2008, at 9:36 AM, Alex Balashov wrote:
>
> 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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