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New Member

Dial Access in SRST

We are using MGCP on our local voice network. We have analog lines as the SRST backup. These lines go into a PBX that is expecting a 9. I have configured the dial-access 9 fxo command. It works, however I have to dial 99xxxxxxx to access the line. Is there a way to prefix a 9 when in SRST mode without switching to 323 dial-peers?

Don

4 REPLIES
New Member

Re: Dial Access in SRST

I don't think this can be done. May be you try different translation rule

New Member

Re: Dial Access in SRST

What about something like this?

dial-peer voice 1 pots

destination-pattern 9.......

prefix 9

port 1/1/0

Caller would dial 95551234

Digits sent would be 995551234

You might want to try different prefixing and see what works best for you

New Member

Re: Dial Access in SRST

To do this I would have to change the router to H.323. I don't want to do that. I want to keep it MGCP.

Don

New Member

Re: Dial Access in SRST

I am running both H.323 and MGCP on our routers. I am not having any problems.

They only time my phones use the H.323 gateway is when they are in Fallback. Other than that they are Pointed to the MGCP Endpoints.

MGCP is primary. H.323 dial peers only are used when the WAN fails. Callmanager controlls the MGCP calls.

here is a sample:

voice-port 1/0/0

!

voice-port 1/0/1

!

voice-port 1/0/2

!

voice-port 1/0/3

!

voice-port 1/0/4

timeouts interdigit 7

timing hookflash-out 50

connection plar opx 5700

description SRST 555-XXXX

!

voice-port 1/0/5

timeouts interdigit 7

timing hookflash-out 50

connection plar opx 5700

description SRST 555-XXXX

!

voice-port 1/0/6

timing hookflash-out 50

!

voice-port 1/0/7

timeouts interdigit 7

timing hookflash-out 50

description Intercom Access

!

mgcp

mgcp call-agent 192.168.50.20 2427 service-type mgcp version 0.1

mgcp dtmf-relay voip codec all mode out-of-band

mgcp rtp unreachable timeout 1000 action notify

mgcp package-capability rtp-package

no mgcp package-capability res-package

mgcp package-capability sst-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp fax t38 inhibit

mgcp rtp payload-type g726r16 static

!

mgcp profile default

!

!

!

dial-peer cor custom

!

!

!

dial-peer voice 1 pots

description access code for intercom

destination-pattern *4

port 1/0/7

forward-digits 0

!

dial-peer voice 2 pots

application mgcpapp

port 1/0/7

!

dial-peer voice 999100 pots

application mgcpapp

port 1/0/0

!

dial-peer voice 999101 pots

application mgcpapp

port 1/0/1

!

dial-peer voice 999102 pots

application mgcpapp

port 1/0/2

!

dial-peer voice 999103 pots

application mgcpapp

port 1/0/3

!

dial-peer voice 999104 pots

application mgcpapp

port 1/0/4

!

dial-peer voice 999105 pots

application mgcpapp

port 1/0/5

!

dial-peer voice 999107 pots

application mgcpapp

port 1/0/7

!

!

call-manager-fallback

ip source-address 192.168.239.1 port 2000

max-ephones 24

max-dn 24

dialplan-pattern 1 123555.... extension-length 4

keepalive 10

default-destination 5700

access-code fxo 9

voicemail 95555555

The Following comes from the CVOICE Coursebook.

MGCP gateway fallback is a feature that improves the reliability of MGCP branch networks. A WAN link connects the MGCP gateway at a remote site to the Cisco CallManager at a central site, which is the MGCP Call agent. If the WAN link fails, the fallback feature keeps the gateway working as an H.323 gateway.

MGCP gateway fallback works in conjunction with the Survivablity Remote Site Telephony(SRST) feature. SRST allows Cisco gateways and routers to manage connections temporarily for Cisco IP Phones shen a connection to the Cisco CallManager is unavailable.

From the first post I assumed you are running Callmanager.

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