You should not need to do anything special for SIP option ping to work with authentication based SIP connections. Here is an example of a configuration we have for one of our sites and it shows the dial-peer in active state.
voice class sip-profiles 10
request INVITE sip-header SIP-Req-URI modify "<redacted> " "spark.co.nz"
request ANY sip-header From modify "10.154.4.10" "10.154.11.189"
request ANY sip-header From modify "From:(.*)(<sip:)\+(.*@).*>" "From: \2\3spark.co.nz>"
response ANY sip-header From modify "From:(.*)(<sip:)\+(.*@).*>" "From: \2\3spark.co.nz>"
request ANY sip-header From modify "From:(.*)(<sip:.*@).*>" "From: \2spark.co.nz>"
response ANY sip-header From modify "From:(.*)(<sip:.*@).*>" "From: \2spark.co.nz>"
request INVITE sip-header To modify "To:(.*)(<sip:.*@).*>" "To: \2spark.co.nz>"
request ANY sip-header To modify "To:(.*)(<sip:.*@).*>" "To: \2spark.co.nz>"
response ANY sip-header To modify "To:(.*)(<sip:.*@).*>" "To: \2spark.co.nz>"
request ANY sip-header Contact modify "Contact:(.*)(<sip:)\+(.*@.*>)" "Contact: \2\3"
response ANY sip-header Contact modify "Contact:(.*)(<sip:)\+(.*@.*>)" "Contact: \2\3"
request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)(<sip:).*@.*>" "P-Asserted-Identity: \<redacted>@spark.co.nz>"
response ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)(<sip:).*@.*>" "P-Asserted-Identity: \<redacted>@spark.co.nz>"
!
voice class sip-options-keepalive 2000
description Used for Service Provider SIP OPTIONS PING
!
!
voice class server-group 2000
ipv4 <redacted> preference 1
ipv4 <redacted> preference 2
!
voice class tenant 2000
registrar dns:spark.co.nz expires 3600
credentials username <redacted> password 7 <redacted> realm spark.co.nz
authentication username <redacted> password 7 <redacted>
no remote-party-id
timers dns registrar-cache 95
sip-server dns:spark.co.nz
connection-reuse
audio forced
bind control source-interface GigabitEthernet0/0/1
bind media source-interface GigabitEthernet0/0/1
no pass-thru content custom-sdp
sip-profiles 10
outbound-proxy dns:gvc-tg0102.spark.co.nz reuse
early-offer forced
!
dial-peer voice 110 voip
description Outbound calls to PSTN
translation-profile outgoing PSTN-OUT
session protocol sipv2
session server-group 2000
destination e164-pattern-map 2000
voice-class codec 10
voice-class sip tenant 2000
voice-class sip options-keepalive profile 2000
dtmf-relay rtp-nte
no vadStatus of dial peers
nzaurt-cube-01#sh dial-peer voice summary
dial-peer hunt 0
AD PRE PASS SESS-SER-GRP\ OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT KEEPALIVE VRF
1000 voip up up 0 syst NA
1010 voip up up map:1 0 syst SESS-SVR-GRP: 1 active NA
100 voip up up 0 syst NA
110 voip up up map:2000 0 syst SESS-SVR-GRP: 2000 active NA
@dijin wrote:
NY4NTEAMS01#sh dial-peer voice summ
dial-peer hunt 0
AD PRE PASS SESS-SER-GRP\ OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT KEEPALIVE VRF
111 voip down down .T 1 syst sip-server NA
300 voip up up 13232383594 1 syst ipv4:10.114.1.28 active NA
311 voip up up .T 1 syst dns:sip1.cy2.omega-t active NA
AFAIKT dial peer 311 shows as up. What is you're specific problem that you have?
I would suggest that you address that question to SolarWind as the router command shows the dial peer to be up.
No I don't. We do not poll this from any management system. The use case we have for SIP option ping to monitor dial peer status does not extend outside of the voice gateways.
