[asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code
Alex Balashov
abalashov at evaristesys.com
Thu Oct 9 09:39:58 CDT 2008
Not offhand / without seeing the Asterisk side.
On Thu, October 9, 2008 10:26 am, Ketema Harris wrote:
> 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
>>
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
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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