Is it by choice that the call uses dial peer 100 for both inbound and outbound destinations?
Looking at the debug on a PC in N++ I see that I missed that line <blush>, not all easy to read debugs on a cell phone.
The debugs are however quite thin in the output of what happens between dial peer 1 is selected as the inbound and dial peer 100 is selected as the outbound and then as the inbound also.
In general the debug provided does not really reveal what is going on. I would recommend you to turn of these debugs and re-post the output from this.
- debug voip ccapi inout
- debug voip dialpeer all (if all is not an option in your IOS you can skip that and just do debug voip dialpeer)
- debug voip translation
This would be my recommendation for configuration. @Maren Mahoney thanks for the vote of confidence. ':-)'
voice class uri CUCM sip host ipv4:<CM CPE 1> ;CPE Subscriber host ipv4:<CM CPE 2> ;CPE Subscriber host ipv4:<CM CPE 3> ;CPE Subscriber ;add as many line as there are CPE nodes in the CM cluster ! voice class e164-pattern-map 1 description E164 Pattern Map for called number to CUCM e164 +<start of DID number range>T ! voice class server-group 1 ipv4 <CM CPE 1> preference 1 ipv4 <CM CPE 2> preference 2 ipv4 <CM CPE 3> preference 3 description Inbound calls from PSTN to CUCM ! voice class sip-options-keepalive 1 description Used for Server Group SIP OPTIONS PING ! trunk group FXO hunt-scheme round-robin both translation-profile incoming PSTN-IN translation-profile outgoing PSTN-OUT ! voice-port 0/0/3 trunk-group FXO input gain 7 !set to match your region output attenuation -4 cptone SA !set to match your region timeouts call-disconnect 1 timeouts wait-release 1 timing hookflash-out 50 !set to match your region description *** local PTT <number> *** bearer-cap Speech caller-id enable ! voice-port 0/1/0 trunk-group FXO input gain 7 !set to match your region output attenuation -4 cptone SA !set to match your region timeouts call-disconnect 1 timeouts wait-release 1 timing hookflash-out 50 !set to match your region description *** local PTT <number> *** bearer-cap Speech caller-id enable ! dial-peer voice 1000 voip description Outbound calls from CUCM translation-profile incoming NOPLUS-IN voice-class codec 1 session protocol sipv2 incoming uri via CUCM dtmf-relay rtp-nte sip-kpml no vad ! dial-peer voice 1010 voip description Inbound calls to CUCM subscribers modem passthrough nse codec g711ulaw session protocol sipv2 session server-group 1 destination e164-pattern-map 1 voice-class codec 1 voice-class sip options-keepalive profile 1 dtmf-relay rtp-nte sip-kpml no vad ! dial-peer voice 100 pots tone ringback alert-no-PI description Inbound calls from PSTN incoming called-number . direct-inward-dial ! dial-peer voice 110 pots trunkgroup FXO tone ringback alert-no-PI description Outbound calls to PSTN destination-pattern 0T !set to match what you use as the breakout number 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 no digit-strip no sip-register !
Please note that this configuration is for using SIP as the control protocol for your voice gateway in CM. Frankly that is a much better option these days than using H.323.
If you anyway want to stick with H.323 this would be the configuration for that.
voice class e164-pattern-map 1 description E164 Pattern Map for called number to CUCM e164 +<start of DID number range>T ! trunk group FXO hunt-scheme round-robin both translation-profile incoming PSTN-IN translation-profile outgoing PSTN-OUT ! voice-port 0/0/3 trunk-group FXO input gain 7 !set to match your region output attenuation -4 cptone SA !set to match your region timeouts call-disconnect 1 timeouts wait-release 1 timing hookflash-out 50 !set to match your region description *** local PTT <number> *** bearer-cap Speech caller-id enable ! voice-port 0/1/0 trunk-group FXO input gain 7 !set to match your region output attenuation -4 cptone SA !set to match your region timeouts call-disconnect 1 timeouts wait-release 1 timing hookflash-out 50 !set to match your region description *** local PTT <number> *** bearer-cap Speech caller-id enable ! dial-peer voice 1000 voip description Outbound calls from CUCM translation-profile incoming NOPLUS-IN voice-class codec 1 incoming called-number . dtmf-relay h245-alphanumeric rtp-nte no vad ! dial-peer voice 1010 voip description Inbound calls to CUCM subscriber 1 preference 1 modem passthrough nse codec g711ulaw session target <CM CPE 1> destination e164-pattern-map 1 voice-class codec 1 dtmf-relay h245-alphanumeric rtp-nte no vad ! ! dial-peer voice 1011 voip description Inbound calls to CUCM subscriber 2 preference 2 modem passthrough nse codec g711ulaw session target <CM CPE 2> destination e164-pattern-map 1 voice-class codec 1 dtmf-relay h245-alphanumeric rtp-nte no vad ! dial-peer voice 1012 voip description Inbound calls to CUCM subscriber 3 preference 3 modem passthrough nse codec g711ulaw session target <CM CPE 3> destination e164-pattern-map 1 voice-class codec 1 dtmf-relay h245-alphanumeric rtp-nte no vad ! dial-peer voice 100 pots tone ringback alert-no-PI description Inbound calls from PSTN incoming called-number . direct-inward-dial ! dial-peer voice 110 pots trunkgroup FXO tone ringback alert-no-PI description Outbound calls to PSTN destination-pattern 0T !set to match what you use as the breakout number 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 no digit-strip no sip-register !
There are other part of the config that is not shown, like translations as such, as these are very specific to your needs. So you'd need to adapt the suggested configurations to match your specific needs.
