Absolutely sound advice, but I would suggest the following alterations to your configuration suggestion.
dial-peer voice 20 voip description ### Incoming from CUCM ### incoming called-number . voice-class codec 1 dtmf-relay h245-alphanumeric h245-signal cisco-rtp no vad ! dial-peer voice 22 voip description ### Outgoing to CUCMPub ### preference 2 destination-pattern 4...$ session target ipv4:10.60.1.35 voice-class codec 1 dtmf-relay h245-alphanumeric h245-signal cisco-rtp no vad huntstop ! dial-peer voice 23 voip description ### Outgoing to CUCMSub ### preference 1 destination-pattern 4...$ session target ipv4:10.60.1.36 voice-class codec 1 dtmf-relay h245-alphanumeric h245-signal cisco-rtp no vad ! dial-peer voice 1 pots description ### Incoming from PSTN ### tone ringback alert-no-PI incoming called-number . direct-inward-dial ! dial-peer voice 10 pots description ### Outgoing to PSTN ### tone ringback alert-no-PI destination-pattern 9T progress_ind setup enable 3 progress_ind alert enable 8 progress_ind progress enable 8 progress_ind connect enable 8 progress_ind disconnect enable 8 port 0/1/0:1 huntstop
@chaitusite Based on the shared configuration and debugs it looks like you have a loop in your call setup and this is caused by that you have overlaps between the match on your dial peers. This is not a good state to have. As advised by @b.winter suggest that you alter the way you send calls from CM to your VGW and how your match the calls outbound to PSTN. If you use a break out code, for example 9, the easist is to keep that in the called number as the call is passed from CM to the gateway. Then in the gateway you use this to match the call going to PSTN and consume that on it's way out to the service provider. There are many ways to achieve this, the simplest is to let the default function of POTS dial peers to consume any explicitly matched digits to be stripped. If you don't want that or have other needs for how to modify the called number you can do this with voice translation rules. Let us know you preference and we can lend you a hand with that part as well.
AFAIKT none of the suggested changes has been implemented. What is you intent with posting your configuration?
Based on your complete shared configuration I would suggest that you make these changes.
! voice service voip no allow-connections h323 to h323 no allow-connections h323 to sip no allow-connections sip to h323 no allow-connections sip to sip ! voice translation-rule 1 rule 1 /^9\(.*\)/ /\1/ ! voice translation-profile PSTN-OUT translate called 1 ! voice-port 0/1/0:1 translation-profile outgoing PSTN-OUT ! dial-peer voice 1 pots description ### Incoming from PSTN ### tone ringback alert-no-PI call-block translation-profile incoming REJECT-CALLS tone ringback alert-no-PI incoming called-number . direct-inward-dial ! dial-peer voice 10 pots description ### Outgoing to PSTN ### tone ringback alert-no-PI destination-pattern 9T progress_ind setup enable 3 progress_ind alert enable 8 progress_ind progress enable 8 progress_ind connect enable 8 progress_ind disconnect enable 8 port 0/1/0:1 huntstop forward-digits all ! dial-peer voice 20 voip description ### Outgoing to CUCMPub ### preference 2 destination-pattern [346]...$ session target ipv4:10.6.X.X voice-class codec 1 dtmf-relay h245-alphanumeric h245-signal cisco-rtp huntstop no vad ! dial-peer voice 21 voip description ### Outgoing to CUCMSub ### preference 1 destination-pattern [346]...$ session target ipv4:10.6.X.X voice-class codec 1 dtmf-relay h245-alphanumeric h245-signal cisco-rtp no vad !
dial-peer voice 11 voip
description ### Incoming from CUCM ###
incoming called-number .
voice-class codec 1
dtmf-relay h245-alphanumeric h245-signal cisco-rtp
no vad
! no dial-peer voice 22 voip no dial-peer voice 23 voip no dial-peer voice 24 voip no dial-peer voice 25 voip
No, those are for if the gateway should act as a Session Border Controller (SBC), or CUBE as Cisco calls it. What these commands do is to enable the gateway to handle VoIP to VoIP call legs, aka two voip dial peers in the call flow. Meaning that you have one voip dial peer match inbound and another, or the same depending on configuration, matched in the outbound direction.
What you have is a traditional TDM (Time Division Multiplex) gateway, meaning that it handles POTS to VoIP, and the reverse, call legs. This is what you have as you have one pots or voip dial peer match inbound, depending if the call comes from PSTN or CM, and then a voip or pots dial peer matched outbound, depending if the call is destined to CM or PSTN. For this none of these commands are needed.
As @Roger Kallberg mentioned, use $ at the end of the destination pattern. Instead of 6 dial peer you can achieve the same with two dial-peer.
use [346]...$
