[cisco-voip] Cisco 2811 SIP trunk and FXS port

[george.hendrix at l-3com.com]

Hi,

 

  I have a remote site with a 2811 router connected to a PSTN via SIP
trunk.  I have a requirement to connect a fax to an FXS/DID port on the
router.  I have it setup somewhat, however, the fxs port can only
receive calls.  If I attempt to place an outbound call, the line simply
goes to a fast busy after several seconds.  Below is an extraction of
the config showing the pstn dial-peers and the fxs port configuration.
Right now, I am just testing this with a regular pots phone.  Any ideas
as to what I am missing or have configured wrong?  Thanks.

 

voice service voip

 allow-connections h323 to h323

 allow-connections h323 to sip

 allow-connections sip to h323

 allow-connections sip to sip

 supplementary-service h450.12

 fax protocol pass-through g711ulaw

 h323

  no h225 timeout keepalive

 sip

  rel1xx disable

!

!

voice class codec 9999

 codec preference 1 g711ulaw

 

voice translation-rule 1

 rule 1 /^30481/ //

!

voice translation-rule 2

 rule 2 /15084/ /3048115084/

!

 

voice translation-profile fax-outgoing

 translate calling 2

!

voice translation-profile incoming

 translate called 1

!

!

voice-card 0

 dspfarm

 dsp services dspfarm

 

voice-port 0/3/0

 station-id number 3048115084

 

 

dial-peer voice 1 voip

 translation-profile incoming incoming

 destination-pattern ..........

 voice-class codec 9999

 voice-class sip profiles 1

 session protocol sipv2

 session target sip-server

 dtmf-relay rtp-nte digit-drop

 dtmf-interworking rtp-nte

 no vad

!

dial-peer voice 11 voip

 translation-profile incoming incoming

 destination-pattern 1..........

 voice-class codec 9999

 voice-class sip profiles 1

 session protocol sipv2

 session target sip-server

 dtmf-relay rtp-nte digit-drop

 dtmf-interworking rtp-nte

!

dial-peer voice 21 voip

 translation-profile incoming incoming

 voice-class codec 9999

 voice-class sip profiles 1

 session protocol sipv2

 session target sip-server

 incoming called-number 304816....

 dtmf-relay rtp-nte digit-drop

 dtmf-interworking rtp-nte

!

dial-peer voice 60 pots

 translation-profile incoming fax-outgoing

 destination-pattern 15084

 direct-inward-dial

 port 0/3/0

!

!

sip-ua

 retry invite 2

 retry bye 2

 retry cancel 2

 sip-server ipv4:10.10.10.1:5152

 

 

Bill Hendrix

L-3 Communications

george.hendrix at l-3com.com <mailto:george.hendrix at l-3com.com> 

EITS Service Desk: 1-800-871-9983

Service Desk email: L-3IT.Help at l-3com.com
<mailto:L-3IT.Help at l-3com.com/omailto:L-3IT.Help at l-3com.com> 

 

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