SIP Gateway - Voice quality

Unanswered Question
Aug 3rd, 2009

Hello All,

I have a media server which runs IVR connected to Voice gateway. Media server runs on SIP and cisco 2811 is configured as SIP gateway. Gateway Connects to E1 pri for DID. I have a problem with voice quality. When we call,gateway redirects to media server and IVR starts but with voice cuts. So is the problem with gateway config or Media server.?

The other problem is medai server can call outside and gateway is configured to do so. But when we pick up the call, it disconnects.


Dial-peer voice 1 pots

description Incoming calls from PSTN

Incoming called-number .T


Port 0/0/1:15

dial-peer voice 2 voip

description Incoming calls to server

destination-pattern 448447.

session protocol sipv2

session target ipv4:{Media Server IP}

incoming called-number 448447.

dtmf-relay rtp-nte

codec g711alaw

Dial-peer voice 3 pots

description Outgoing

destination-pattern 0[7-8]0........

Port 0/0/1:15

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

redirect ip2ip

signaling forward unconditional

fax protocol cisco



interface FastEthernet0/1

description $WAN INTERFACE$

ip address

ip nat outside

ip virtual-reassembly

no ip mroute-cache

duplex auto

speed auto


interface Serial0/0/0:15

no ip address a.b.c.d

encapsulation hdlc

isdn switch-type primary-net5

isdn incoming-voice voice

no cdp enable

I have this problem too.
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Nicholas Matthews Mon, 08/03/2009 - 06:14

Try a few things:

network-clock-select 1 e1 0/0/0


dial-peer voice 2 voip

incoming called-number .




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