How to connect a Cisco Router with FXO module to Asterisk Cisco routers support FXO or FXS voice interface

cards (aka VIC). Either SIP or H.323 can be used for call signalling. VICs are installed into low density voice modules (NM-1V, or NM-2V) in a Cisco 2600/3600/3700 router. Data Sheet. * REGISTER: Cisco gateway doesn't usually REGISTER with SIP servers, but lat er versions of IOS can (in order to register FXS ports, for instance) * Codecs: Cisco gateway do not support GSM, iLBC & Speex CODECs. Prior to Ci sco IOS Software Release 12.0(5)T, VoIP gateways only supported the G.729 and G. 711 codecs and only one voice call per DSP. Version 12.0(5)T has added support f or G.723 and G.726, and now support up to four voice/fax-relay calls per DSP. Mo re on Cisco DSPs. Here are the codecs Cisco offers at the router: clear-channel Clear Channel 64000 bps g711alaw G.711 A Law 64000 bps g711ulaw G.711 u Law 64000 bps g723ar53 G.723.1 ANNEX-A 5300 bps g723ar63 G.723.1 ANNEX-A 6300 bps g723r53 G.723.1 5300 bps g723r63 G.723.1 6300 bps g726r16 G.726 16000 bps g726r24 G.726 24000 bps g726r32 G.726 32000 bps g728 G.728 16000 bps g729br8 G.729 ANNEX-B 8000 bps g729r8 G.729 8000 bps gsmefr GSMEFR 12200 bps gsmfr GSMFR 13200 bps Sample configuration sip.conf [general] port=5060 bindaddr=0.0.0.0 videosupport=yes ; Gives an error "process_sdp: Error in codec string 'ideo 0'" disallow=all allow=ulaw context=bogon-calls [y.y.y.y] context=pstn-incoming type=friend host=y.y.y.y ; IP address of Cisco gateway dtmfmode=rfc2833 disallow=all allow=ulaw [1001] context=local-phones type=friend username=1001 secret=secret host=dynamic mailbox=1001

extensions.conf [bogon-calls] exten => _.,1,Congestion [pstn-incoming] include => lan-phones [local-phones] include => lan-phones include => pstn-outbound [pstn-outbound] ; Calls starting with 9 have the 9 stripped & are then routed out to the PSTN exten => _9.,1,Dial(SIP/${EXTEN 1}@y.y.y.y) ; IP address of Cisco gateway ; 9 stripped by Cisco gateway exten => _9XXXX,1,Dial,SIP/${EXTEN}@y.y.y.y ; IP address of Cisco gateway exten => _9XXXX,2,Congestion [lan-phones] exten => 1001,1,Dial(SIP/1001,20) exten => 1001,2,Voicemail(u1001) exten => 1001,102,Voicemail(b1001) exten => 1001,103,Hangup Cisco config SIP requires routers to be set to use GMT clock timezone GMT 0 voice-port 1/0/0 We are in the UK cptone GB input gain 10 output attenuation 10 no comfort-noise Route all calls into this FXO port on to extension 1001 connection plar opx 1001 PSTN gateway (PBX) 9 is stripped dial-peer voice 100 pots destination-pattern 9.... The above line will match anything passed from * to the Cisco starting with a 9, the dial peer then strips the 9 and passes it to the FXO port If you want to match any dialed number from * starting with 9, use destination-p attern 9T port 1/0/0 forward-digits 4 dial-peer voice 2 voip description Route calls starting with 1 to the Asterisk PBX destination-pattern 1... session protocol sipv2 x.x.x.x = IP address of * session target ipv4:x.x.x.x:5060 session transport udp dtmf-relay rtp-nte codec g711ulaw

if clid is present then Asterisk tries to mtach that as a user clid strip no vad

sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 x.x.x.x = IP address of * sip-server ipv4:x.x.x.x If using G729: * IOS 12.4 needs the flawed g729br8 * May need to specify packet size: codec g729r8 bytes 40 Caller ID The VIC-2FXO voice interface card does not support Caller ID. In order to suppor t that, you will need battery reversal support (in the US, VIC-2FXO-M1; in Europ e, VIC-2FXO-M2, etc.). More on Cisco FXO interfaces: http://cisco.com/en/US/prod ucts/hw/routers/ps274/products_tech_note09186a00800b53c7.shtml If you have a VIC-2FXO-M1 and wish to have Caller ID information passed through to Asterisk, you need to set up a null translation rule for that voice port. For example: translation-rule 1 rule 1 null null voice-port 1/0/0 translate calling 1 translate called 1 caller-id enable This will permit Caller ID information to be passed through to the SIP connectio n. Another Take on Caller ID I may have inadvertently backed into a solution on this issue. I previously had figured out how to make the VIC-2FXO (not VIC-2FXO-M1) work with CallManager 4.1 . The issue there was that CLID didn't work with MGCP, but research lead me to a solution that did work for that card with H.323. So, when I converted these por ts to SIP for Asterisk, I found that they passed caller ID just fine! Here is th e relevant configuration: voice-port 1/0/0 output attenuation -1 no comfort-noise connection plar opx 7145551212 description 71455551212 caller-id enable dial-peer voice 1 pots incoming called-number . destination-pattern . port 1/0/0

dial-peer voice 100 voip preference 1 incoming called-number . destination-pattern 7145551212 progress_ind setup enable 3 session protocol sipv2 session target ipv4:x.x.x.x:5060 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:x.x.x.x I haven't yet determined if all of this is in fact required, just that we are us ing this configuration (with our real numbers of course) and we see caller ID ju st fine. I hope this helps someone! The IOS version in this case was 12.3(21), w hich may be an alternate explanation for why various folks say it won't work and yet it in fact did. # NOTE *** If you are using asterisk 1.6 and are having trouble with outbound PSTN calls (t hey drop before connection is made), please see my comment in the comments secti on (added by 'ahhyes')