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 >>> >>> >>> _______________________________________________ >>> -- 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 >
-- 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
