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 > > > _______________________________________________ > -- 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 _______________________________________________ -- 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
