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 >> >> >> _______________________________________________ >> -- 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
