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Why Dial Peer Isn't Being Matched

Michael Mertens
Level 1
Level 1

I have an existing CUBE with a functioning SIP trunk. I am provisioning a 2nd SIP trunk on the CUBE and am trying to test outbound calling on that new SIP trunk by creating a new/temporary dial-peer (dialpeer 210) with a destination-pattern of my cell phone 518-555-1212; however, I never seem to be able to match that and continue to match the existing outbound dial-peer (dialpeer) with destination 9T. I even tried to change the preference of both dial-peer 201 and 210, but I continue to match on 201. I thought that outbound dialpeer matching would be based on closest match, or is the T in the 9T pattern making an instance decision? Is there a better way for me to force a match on 210? Please see below and THANKS!!!!

 

# Show call active voice br | in pid

0 : 2951124 3941132450ms.1 (13:35:33.525 UTC Fri Nov 4 2022) +-1 pid:100 Answer 26468 connecting
0 : 2951126 3941133510ms.1 (13:35:34.585 UTC Fri Nov 4 2022) +-1 pid:202 Originate 15185551212 connected

!INCOMING CALL LEG
dial-peer voice 100 voip
description Incoming from CallManager
session protocol sipv2
incoming called-number 9T
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad


! OUTGOING CALL LEG WHICH KEEPS MATCHING
dial-peer voice 201 voip
description EXISTING Outgoing to AT&T -AT&T Call Leg Primary IPBE
translation-profile outgoing pstn-out
preference 1
destination-pattern 9T
no modem passthrough
session protocol sipv2
session target ipv4:12.194.41.30
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
ip qos dscp cs3 signaling
no vad
!
!
dial-peer voice 210 voip
description ONSEMI Outgoing to AT&T -AT&T Call Leg Quarternary IPBE
translation-profile outgoing onsemi-pstn-out
preference 1
destination-pattern 915185551212
no modem passthrough
session protocol sipv2
session target ipv4:12.253.102.70
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
ip qos dscp cs3 signaling
no vad
!

1 Accepted Solution

Accepted Solutions

TechLvr
Spotlight
Spotlight

In this case your outgoing call leg matched dial peer 202 which you have not included in your post. 
Please share all your dial peers for better analysis. 

# Show call active voice br | in pid
0 : 2951124 3941132450ms.1 (13:35:33.525 UTC Fri Nov 4 2022) +-1 pid:100 Answer 26468 connecting
0 : 2951126 3941133510ms.1 (13:35:34.585 UTC Fri Nov 4 2022) +-1 pid:202 Originate 15185551212 connected

Also, since dial peer 210 uses a different target session make sure the IP address is reachable/functional. 
session target ipv4:12.253.102.70

View solution in original post

1 Reply 1

TechLvr
Spotlight
Spotlight

In this case your outgoing call leg matched dial peer 202 which you have not included in your post. 
Please share all your dial peers for better analysis. 

# Show call active voice br | in pid
0 : 2951124 3941132450ms.1 (13:35:33.525 UTC Fri Nov 4 2022) +-1 pid:100 Answer 26468 connecting
0 : 2951126 3941133510ms.1 (13:35:34.585 UTC Fri Nov 4 2022) +-1 pid:202 Originate 15185551212 connected

Also, since dial peer 210 uses a different target session make sure the IP address is reachable/functional. 
session target ipv4:12.253.102.70