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New Member

Call preservation with MGCP fallback possible?

I got a MGCP gateway w/ SRST and a T1 PRI trunk for PSTN calls. When I lost the WAN, SRST worked but I faced 2 issues:

1. Active calls didn't survive, and

2. I couldn't make outbound calls

I think I have some workarounds for the 2nd issue. However, the 1st issue is what I really wanted to get to the bottom of it. I know if I lose all callmanagers, PRI will get backhauled to the SRST router, then D-channel gets reset and my call will be dropped. Is there a way to fix this problem? I want to keep MGCP if possible.

4 REPLIES
New Member

Re: Call preservation with MGCP fallback possible?

MGCP does not support call pres with ISDN PRI or

BRI

It will work for T1 CAS and POTS

H323 call pres is the way to go. I would recommend to all only use H323 or SIP trunking. You can control you call routing much cleaner. I know it is more configuration on the front end of deployment but so many applications coming out rely on you GW to be SIP or H323. Especially if you are planing contact center at any time in you solution.

You answered your own question in your post the D Channel drops. There is no way to get around that with out going to a diffrent protocol. even with MGCP fall back.

Why are you stuck on MGCP when we all know that protocol is going away. Sorry Cisco I know you do not agree with me but I saw the writing on the wall in The 2000 's when Megaco changed to MGCP and still had to many caveauts.

Cisco Employee

Re: Call preservation with MGCP fallback possible?

This topic has been covered before. See below,

https://supportforums.cisco.com/docs/DOC-3048

-Felipe

New Member

Re: Call preservation with MGCP fallback possible?

OK I read the topic from that link before. However, I just want to confirm if I got it right. So what I would need to do is:

From the CUCM:

-  Remove MGCP gw

-  Do H.323 trunk to the SRST router

-  Make sure all off-net calls point to the SRST router which I will be converting from MGCP to H.323, i.e. the route pattern piece

From the SRST router:

-  Remove MGCP with a "no mgcp" command and anything mgcp related.

-  Put in H.323 config like pots and voip dial-peers, etc.

So does that sound about right? I wanted to try it tonight. It's been awhile I haven't touched H.323 so things might break when I switch

I just need some core config to make it work first. So if you guys got any doc or URL to start off with. That'd be greatly appreciated. Thanks.

New Member

Re: Call preservation with MGCP fallback possible?

Please check and confirm if this config is good or not. Thanks.

!

voice service voip

h323

  call preserve

  no h225 timeout keepalive

!

voice class h323 1

h225 timeout tcp establish 3

!

application

global

  service alternate default

!

interface FastEthernet0/0

ip address 10.10.10.6 255.255.255.0

h323-gateway voip bind scraddr 10.10.10.6

duplex auto

speed auto

!

voice-card 0

dsp services dspfarm

!

network-clock-participate wic 1

network-clock-select 1 T1 0/1/0

!

controller T1 0/1/0

framing esf

linecode b8zs

cablelength long 0db

pri-group timeslots 1-24

!

interface Serial0/1/0:23

no ip address

encapsulation hdlc

isdn switch-type primary-ni

isdn incoming-voice voice

no cdp enable

!

###### HERE ARE SOME TYPICAL DIAL PEERS #####

!

dial-peer voice 6000 voip

preference 1

destination-pattern 6...

voice-class h323 1

session target ipv4:20.20.20.10

incoming called-number 24..

dtmf-relay h245-alphanumeric

codec g711ulaw

ip qos dscp cs3 signaling

no vad

!

dial-peer voice 9 pots

description Default Outbound PSTN calls

preference 1

destination-pattern 9T

incoming called-number 24..

direct-inward-dial

port 0/1/0:23

!

dial-peer voice 911 pots

description ** 911 Calls **

destination-pattern 911

port 0/1/0:23

forward-digits all

!

call-manager-fallback

secondary-dialtone 9

max-conferences 4 gain -6

transfer-system full-consult

limit-dn 7910 1

limit-dn 7935 1

limit-dn 7940 2

limit-dn 7960 2

limit-dn 7970 2

ip source-address 10.10.10.6 port 2000

max-ephones 30

max-dn 60 dual-line

system message primary SRST Fallback Local

dialplan-pattern 1 XXX32624.. extension-length 4 !!!!!!!!!!!

voicemail 91XXX1234567

call-forward busy 91XXX1234567

call-forward noan 91XXX1234567 timeout 15

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