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

[Nick Matthews]

There are probably 20 things that could cause this, only few of which
may be inferred from the configuration.  'debug ccsip messages' is
much better for determining the problem.

-nick

On Thu, Nov 19, 2009 at 1:29 PM,  <george.hendrix at l-3com.com> wrote:
> 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
>
> EITS Service Desk: 1-800-871-9983
>
> Service Desk email: L-3IT.Help at l-3com.com
>
>
>
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