09-17-2010 01:57 AM - edited 03-16-2019 12:50 AM
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.
09-17-2010 06:38 AM
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.
09-11-2023 11:27 AM - edited 09-11-2023 11:27 AM
13 years later and MGCP still going strong.
09-11-2023 01:07 PM
I wouldn’t say that. It’s in my experience not very common any longer. SIP has since long become the de-facto standard.
09-17-2010 06:38 AM
09-17-2010 08:56 AM
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.
09-17-2010 12:41 PM
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
09-28-2023 08:11 AM - edited 09-30-2023 06:18 AM
During the WAN failure and the subsequent switch to SRST, my active calls didn't survive the transition and got dropped. I'm keen to understand if there's a way to ensure that these active calls persist during failover. Maintaining call continuity is crucial for our operations, and I'd like to explore solutions that enable this while still using MGCP if possible. I encountered problems with making outbound calls during SRST mode.
09-28-2023 08:22 AM
The short answer is that MGCP does not support call preservation with ISDN due to the reset of the D-Channel. If your circuits are POTS or T1-CAS you are able configure call preservation during MGCP Fallback. AFAIK there is no getting around the ISDN problem with MGCP fallback.
Maren
09-28-2023 09:28 AM
Did you not read the previous answers on this post? It’s clearly stated that call preservation is not supported with MGCP.
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