11-04-2022 06:58 AM
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
!
Solved! Go to Solution.
11-04-2022 07:13 AM - edited 11-04-2022 07:33 AM
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
11-04-2022 07:13 AM - edited 11-04-2022 07:33 AM
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
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide