06-05-2013
07:44 AM
- last edited on
03-25-2019
08:23 PM
by
ciscomoderator
Yesterday, I was able to get outbound calls up and running with the help of the community here, thanks again! Thanks to anyone willing to give a hand with the inbound calls!
I am unable to receive incoming calls on phones registered to my CUCM cluster. I have a SIP trunk from my CUBE to my ITSP, and another SIP trunk from CUBE to CUCM. I am able to place outbound calls from phones registered to CUCM out to the PSTN via my ITSP, and I am also able to receive calls to phones from my ITSP if they are registered to CUCME on my CUBE. I just can't get them working all the way from my ITSP to CUCM. Does anyone have any ideas?
Configurations are attached for anyone who might have some ideas. I've replaced real PSTN-facing phone numbers with ******** for privacy reasons. Thank you in advance for any help you might be able to offer!
voice translation-rule 1
rule 1 /616*******/ /1003/
voice translation-rule 2
rule 1 /^9\(.......\)$/ /616\1/
rule 2 /^91\(..........\)/ /1\1/
voice translation-profile Incoming_DID
translate called 1
!
voice translation-profile Outgoing_Calls
translate called 2
dial-peer voice 1 voip
description Dial Peer to receive incoming phone calls from ITSP
translation-profile incoming Incoming_DID
voice-class sip asserted-id ppi
session target dns:chicago.voip.ms
incoming called-number 616*******
dtmf-relay rtp-nte
!
dial-peer voice 2 voip
description Outbound Local Dialing
translation-profile outgoing Outgoing_Calls
destination-pattern 9[^1]......
session protocol sipv2
session target dns:chicago.voip.ms
session transport udp
dtmf-relay rtp-nte
clid network-number 616*******
!
dial-peer voice 5 voip
description Outbound Long Distance Dialing
translation-profile outgoing Outgoing_Calls
destination-pattern 91..........
session protocol sipv2
session target dns:chicago.voip.ms
session transport udp
dtmf-relay rtp-nte
clid network-number 616*******
!
dial-peer voice 1000 voip
description Production Phones Registered to CUCM
destination-pattern 100.
session protocol sipv2
session target ipv4:10.1.40.240
dtmf-relay rtp-nte sip-notify
codec g711ulaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 3 voip
description Local Dialing between CUCM to ITSP
session protocol sipv2
session target dns:10.1.40.240
session transport udp
incoming called-number 9[^1]......
dtmf-relay rtp-nte
!
dial-peer voice 4 voip
description Long Distance Dialing between CUCM to ITSP
session protocol sipv2
session target dns:10.1.40.240
session transport udp
incoming called-number 91..........
dtmf-relay rtp-nte
06-05-2013 09:05 AM
Hi again,
can you tell us in what format - I don't want you to show the whole thing, just the first two-three digits - does the ITSP send the called number? Is it 616 something something?
This way: the incoming-called number must match - well, the called number. Of couse, you can do translations afterwards, but the router needs to have something to relate to, a more or less fixed point in the universe.
For a simple test, could you rephrase your dial-peer 1 like this:
dial-peer voice 1 voip
description Dial Peer to receive incoming phone calls from ITSP
! let's not translate anything for now
voice-class sip asserted-id ppi
! no need for session-target, it's used for outbound dp only
incoming called-number .
dtmf-relay rtp-nte sip-notify
Would it be possible to have a testing "window" when no other calls are present, just to eliminate the noise?
If yes, can you do deb voip dialpeer and then deb ccsip eve and post the result to here?
G.
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