[cisco-voip] SIP calls problem

[Ratko Dodevski]

This is my situation; I have a Cisco VG 2801 whit FXS and BRI ports.
Currently I have analogue phone connected in one of the FXS ports. This VG
needs to connect to softswitch using SIP UA. IN front of the softswitch
there is a SBC (session border controller) which handles the calls.I managed
to create connection and made them to negotiate same codec and use the SBC
as SIP proxy, but I still don’t hear anything. In the attach there is the
current configuration and a test call from and to the phone on the gateway.
Also, I see on the softswitch that when I try to reach a number registered
on the softswitch I get SIP message 488 Not Acceptable Here. I should
mention that calls to other networks (PSTN, GSM) are just fine, problem is
just with the calls to phones on the softswitch. Also, this is a production
network, and the VG is the only new component. Also calls to VG from any
network (including VoIP phones)are working.

Only Difference when debugging is bridge done

        Preferred Codec        : g729r8, bytes :20

        Preferred  DTMF relay  : rtp-nte

        Preferred NTE payload  : 101

        Early Media            : No

        Delayed Media          : No

        Bridge Done            : *No*

        New Media              : No

        DSP DNLD Reqd          : No

When I receive a call Bridge Done is *NO* and when I make a call is *YES*.

What does this means? Is it something relevant?


                                             |-----VoIP_Phones

                                             |

Analogue_Phone-----(FXS)CiscoVG(SIP)--------SBC----------SoftSwitch

                                             |

                                             |-----------PSTN

                                             |

                                             |-----------GSM


Can anyone provide me with some help or share an expirience?

-- 
Ratko
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