04-30-2020 01:19 AM
Dears, hope you all doing well,
We have this environment (call flow):
ITSP (SIP Trunk) --> CUBE --> CUCM (5001 is AA) --> CUC (Customer Service 0 is a hunt group on CUCM extension number 1888)
We have one way-audio and the call will disconnect after 10-15 seconds when calling to this hunt-group, the weird thing is when I do a "debug ccsip messages" the call will work fine
This is the running config on my CUBE
! voice-card 0 dspfarm dsp services dspfarm ! ! voice rtp send-recv ! voice service voip ip address trusted list ipv4 172.25.227.29 255.255.255.255 ipv4 10.141.40.233 255.255.255.255 ipv4 10.170.16.69 255.255.255.255 mode border-element allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip signaling forward unconditional fax protocol pass-through g711alaw sip early-offer forced no call service stop ! voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729r8 codec preference 4 g729br8 ! ! voice class dualtone-detect-params 1 freq-max-power 0 freq-min-power 13 freq-power-twist 4 cadence-variation 8 ! ! voice class e164-pattern-map 1 description ##ALL INCOMING NUMBERS FROM STC## e164 ^01142087.. e164 ^42087.. ! ! voice class e164-pattern-map 2 description ##INCOMING NUMBERS FROM CUCM## e164 ^01142087.. e164 ^42087.. ! ! voice class server-group 1 ipv4 172.25.227.29 preference 1 description ##CUCM## ! ! ! ! ! voice translation-rule 2 rule 1 /^.*/ /5001/ ! ! voice translation-rule 4 rule 1 /^9/ // ! ! voice translation-profile IN-FROM-STC translate called 2 ! voice translation-profile OUT-TO-STC translate called 4 ! ! ! ! ! ! ! interface GigabitEthernet0/0 description ## LAN ## ip address 172.25.227.126 255.255.255.0 duplex auto speed auto ! ! interface GigabitEthernet0/2 description ## STC_SIP_CIRCUIT ## ip address 10.170.16.70 255.255.255.252 duplex auto speed auto ! ! no ip http server no ip http secure-server ! ip route 10.141.40.233 255.255.255.255 10.170.16.69 ip ssh version 2 ! ! mgcp behavior rsip-range tgcp-only mgcp behavior comedia-role none mgcp behavior comedia-check-media-src disable mgcp behavior comedia-sdp-force disable ! mgcp profile default ! ! ! ! dspfarm profile 1 transcode universal codec g729abr8 codec g729ar8 codec g711alaw codec g711ulaw codec g729r8 codec g722-64 maximum sessions 10 associate application CUBE ! dial-peer voice 1 voip description ##ALL INCOMING NUMBERS FROM ITSP## translation-profile incoming IN-FROM-STC session protocol sipv2 session target ipv4:172.25.227.29 session transport udp incoming called e164-pattern-map 1 voice-class codec 1 voice-class sip bind control source-interface GigabitEthernet0/2 voice-class sip bind media source-interface GigabitEthernet0/2 dtmf-relay rtp-nte sip-kpml no fax-relay sg3-to-g3 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none no vad ! dial-peer voice 2 voip description ##TO CUCM## destination-pattern 5001 session protocol sipv2 session transport udp session server-group 1 voice-class codec 1 voice-class sip bind control source-interface GigabitEthernet0/0 voice-class sip bind media source-interface GigabitEthernet0/0 dtmf-relay rtp-nte sip-kpml fax-relay ecm disable no fax-relay sg3-to-g3 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none no vad ! dial-peer voice 3 voip description ##ALL INCOMING NUMBERS FROM CUCM## session protocol sipv2 session transport udp incoming calling e164-pattern-map 2 voice-class codec 1 voice-class sip bind control source-interface GigabitEthernet0/0 voice-class sip bind media source-interface GigabitEthernet0/0 dtmf-relay rtp-nte fax-relay ecm disable no fax-relay sg3-to-g3 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none no vad ! dial-peer voice 4 voip description ## OUTGOING CALL TO ITSP## translation-profile outgoing OUT-TO-STC destination-pattern 9T session protocol sipv2 session target ipv4:10.141.40.233 session transport udp voice-class codec 1 offer-all voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/2 voice-class sip bind media source-interface GigabitEthernet0/2 dtmf-relay rtp-nte cisco-rtp h245-alphanumeric h245-signal sip-kpml sip-notify no vad !
Thanks in advance
Solved! Go to Solution.
05-03-2020 07:46 AM
That’s awesome. Thanks for the feedback that it’s working for you now. Glad to be able to help you out.
05-05-2020 01:34 AM
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