matthnick at gmail
Nov 20, 2009, 12:54 PM
Post #2 of 2
(1437 views)
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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> 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
>
> EITS Service Desk: 1-800-871-9983
>
> Service Desk email: L-3IT.Help [at] l-3com
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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